diff options
author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-17 16:01:51 +0000 |
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committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-17 16:01:51 +0000 |
commit | 62690f65202eb8b127a8c3880c42591de7087917 (patch) | |
tree | 931e47a077dc7d27b90d69fc73ad2645639474da /ChangeLog | |
parent | 93f22503251d3bb8c4a25226db788510c5cc51ad (diff) | |
parent | ac96ad37e08f6017a51b29d849e82c6fb98d5782 (diff) |
Creating tag for the release of asterisk-1.6.0.16-rc1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.16-rc1@219206 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'ChangeLog')
-rw-r--r-- | ChangeLog | 53335 |
1 files changed, 0 insertions, 53335 deletions
diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 2986fee1d..000000000 --- a/ChangeLog +++ /dev/null @@ -1,53335 +0,0 @@ -2009-09-17 Leif Madsen <lmadsen@digium.com> - - * Released Asterisk 1.6.0.16-rc1 - -2009-09-16 23:52 +0000 [r219064] Tilghman Lesher <tlesher@digium.com> - - * main/config.c, configs/extensions.conf.sample, /: Merged - revisions 219061 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) - | 15 lines Merged revisions 219023 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) - | 8 lines Properly deal with quotes in the arguments of '#exec' - includes. (closes issue #15583) Reported by: pkempgen Patches: - 20090726__issue15583.diff.txt uploaded by tilghman (license 14) - 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license - 169) Tested by: pkempgen ........ ................ - -2009-09-16 19:26 +0000 [r218935] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | - mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 - lines Reverse order of args to fread. This way, we don't always - write a null byte into byte 1 of the buffer (closes issue #15905) - Reported by: ebroad Patches: freadfix.patch uploaded by ebroad - (license 878) Tested by: ebroad ........ - -2009-09-16 18:44 +0000 [r218931] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | - file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On - TCP and TLS connections do not attempt to stop retransmission of - the packet internally. This was preventing responses from being - properly processed because the packet was not being found causing - handle_response to return prematurely. ........ - -2009-09-16 18:11 +0000 [r218869] David Brooks <dbrooks@digium.com> - - * main/pbx.c, /: Merged revisions 218868 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) - | 20 lines Merged revisions 218867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) - | 13 lines Fixes CID pattern matching behavior to mirror that of - extension pattern matching. Pattern matching for extensions uses - a type of scoring system, giving values for specificity to each - character in the pattern. Unfortunately, this is done character - by character, in order. This does lead to some less specific - patterns being first in line for matching, but it will usually - get the job done. This patch merely brings CID matching to the - same level as extension matching. This patch does not attempt to - tackle the problem shared by extension matching. (closes issue - #14708) Reported by: klaus3000 ........ ................ - -2009-09-16 13:36 +0000 [r218800] Russell Bryant <russell@digium.com> - - * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged - revisions 218799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) - | 16 lines Merged revisions 218798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) - | 9 lines Remove the IAXy firmware from Asterisk. The firmware - can now be found on downloads.digium.com, where the rest of our - binary downloads live. This was the last part of our Asterisk - tarballs that was considered non-free by Debian. :-) (closes - issue #15838) Reported by: paravoid ........ ................ - -2009-09-15 22:39 +0000 [r218732] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 - (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) - | 6 lines If the user enters the same password as before, don't - signal an error when the change does nothing. (closes issue - #15492) Reported by: cbbs70a Patches: - 20090713__issue15492.diff.txt uploaded by tilghman (license 14) - ........ ................ - -2009-09-15 19:31 +0000 [r218690] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | - dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines - upward bound checking for port string to int conversion ........ - -2009-09-15 16:21 +0000 [r218601] Matthew Nicholson <mnicholson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep - 2009) | 15 lines Merged revisions 218578 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep - 2009) | 8 lines Send request contact header field with response - to registrer queries instead of the address of record. (closes - issue #14438) Reported by: ravindrad Patches: regquerypatch - uploaded by ravindrad (license 684) Tested by: ravindrad ........ - ................ - -2009-09-15 16:05 +0000 [r218580] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_followme.c: Merged revisions 218579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) - | 16 lines Merged revisions 218577 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) - | 9 lines Ensure FollowMe sets language in channels it creates. - Also, not in the original bug report, but related fields are - accountcode and musicclass, and the inheritance of datastores. - (closes issue #15372) Reported by: Romik Patches: - 20090828__issue15372.diff.txt uploaded by tilghman (license 14) - Tested by: cervajs ........ ................ - -2009-09-15 15:42 +0000 [r218505-218573] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | - mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 - lines Use a better method of ensuring null-termination of the - buffer while reading the SDP when using TCP. ........ - - * /, channels/chan_sip.c: Merged revisions 218499,218504 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, - 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent - over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 - -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP - socket is null-terminated. ........ - -2009-09-15 15:03 +0000 [r218501] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/Makefile: Merged revisions 218500 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep - 2009) | 9 lines Merged revisions 218497 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep - 2009) | 1 line Use proper hostname for downloading sound files. - ........ ................ - -2009-09-14 22:49 +0000 [r218431] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) - | 18 lines Merged revisions 218401 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) - | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent - crash in do_monitor. After talking to rmudgett about some of his - recent iflist locking changes, it was determined that the only - place that would destroy a channel without being explicitly to do - so was in handle_init_event. The loop to walk the interface list - has been modified to wait to destroy the channel until the - dahdi_pvt of the channel to be destroyed is no longer needed. - (closes issue #15378) Reported by: samy ........ ................ - -2009-09-14 19:49 +0000 [r218362] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /, configs/voicemail.conf.sample, - sounds/Makefile: Merged revisions 218361 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) - | 11 lines Recorded merge of revisions 218331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) - | 4 lines Don't say "Please try again" if we don't give the user - another chance to try again. (issue #15055, SWP-129) Reported by: - jthurman ........ ................ - -2009-09-14 15:22 +0000 [r218244] Matthew Nicholson <mnicholson@digium.com> - - * /, apps/app_directed_pickup.c: Merged revisions 218224 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 - (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep - 2009) | 8 lines Ensure we don't pickup ourselves when doing - pickup by exten. (closes issue #15100) Reported by: lmsteffan - Patches: (modified) pickup.patch uploaded by lmsteffan (license - 779) ........ ................ - -2009-09-13 19:10 +0000 [r218216] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0 - that annoys gcc This memset doesn't write beyond the end of the - buffer. (tmpbuf has size of 4). Merged revisions 218184 via - svnmerge from http://svn.digium.com/svn/asterisk/trunk - -2009-09-12 13:10 +0000 [r218108] Michiel van Baak <michiel@vanbaak.info> - - * main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'. - Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this - patch is different from the fix that went into trunk (closes - issue #15711) Reported by: davidw Patches: - 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) - Tested by: davidw - -2009-09-11 05:58 +0000 [r217920-218051] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 217990 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) - | 10 lines Merged revisions 217989 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) - | 3 lines Don't ring another channel, if there's not enough time - for a queue member to answer. (Fixes AST-228) ........ - ................ - - * contrib/scripts/iax-friends.sql, /, channels/chan_sip.c, - channels/chan_iax2.c: Merged revisions 217916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | - tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines - Make calltoken support work with realtime users and peers. - ........ - -2009-09-10 22:31 +0000 [r217858-217913] David Vossel <dvossel@digium.com> - - * channels/chan_sip.c: sip peer matching by address only with - TCP/TLS This patch removes the contact header matching logic and - adds logic to match all tcp/tls connections by ip only. Thanks to - oej for finding the issue and suggesting solutions. Review: - https://reviewboard.asterisk.org/r/355/ - - * /, channels/chan_iax2.c: Merged revisions 217807 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 - (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) - | 22 lines IAX2 encryption regression The IAX2 Call Token - security patch inadvertently broke the use of encryption due to - the reorganization of code in the socket_process() function. When - encryption is used, an incoming full frame must first be - decrypted before the information elements can be parsed. The - security release mistakenly moved IE parsing before decryption in - order to process the new Call Token IE. To resolve this, - decryption of full frames is once again done before looking into - the frame. This involves searching for an existing callno, - checking the pvt to see if encryption is turned on, and - decrypting the packet before the internal fields of the full - frame are accessed. (closes issue #15834) Reported by: karesmakro - Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel - (license 671) Tested by: dvossel, karesmakro Review: - https://reviewboard.asterisk.org/r/355/ ........ ................ - -2009-09-10 19:53 +0000 [r217736] mnick <mnick@localhost>: - - * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) | - 17 lines Sets the correct musicclass after an announcement - (closes issue #15279) Reported by: mbeckwell Patches: patch.txt - uploaded by mnick (license ) Tested by: mnick (closes issue - #15832) Reported by: mbeckwell Patches: patch.txt uploaded by - mnick (license 874) Tested by: mnick ........ - -2009-09-10 12:16 +0000 [r217596] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | - oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines - Include ActionID in all events that are responsed to AMI Action - SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy - Patches: manager_SIPshowregistry_actionid.patch uploaded by nic - bellamy (license 299) ........ - -2009-09-09 20:15 +0000 [r217484] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc - 4.4 has more strict rules for aliasing. It doesn't like a struct - sockaddr_in pointer pointing to a struct sockaddr. So we make it - a union. Merged revisions 217445 via svnmerge from - http://svn.digium.com/svn/asterisk/trunk - -2009-09-09 11:33 +0000 [r217405] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | - oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not - having any TLS session to write to is a serious XMIT_ERROR. - ........ - -2009-09-08 21:45 +0000 [r217281] Kevin P. Fleming <kpfleming@digium.com> - - * configure: Commit regenerated configure script that I missed - earlier. - -2009-09-08 20:31 +0000 [r217209] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) - | 14 lines Merged revisions 217156 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) - | 7 lines When MOH is playing on the channel, announcements sent - through the conference are not heard. (closes issue #14588) - Reported by: voipas Patches: 20090716__issue14588__2.diff.txt - uploaded by tilghman (license 14) Tested by: lmadsen, twisted, - tilghman ........ ................ - -2009-09-08 16:38 +0000 [r217075] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/autoconfig.h.in, configure.ac: Merged - revisions 217074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | - kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 - lines Ensure that the default autoconf CFLAGS are not used. A - recent change to the configure script that allows the user to - specify CFLAGS and/or LDFLAGS to the script had the unfortunate - side effect of letting autoconf's default CFLAGS (-g -O2) feed in - to the rest of the build system, thereby overriding the - DONT_OPTIMIZE setting in menuselect. That problem is now - corrected. ........ - -2009-09-08 15:35 +0000 [r217034] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_limit.c: Merged revisions 217033 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 | - tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines - Remove what appears to be an unnecessary define. (closes issue - #15851) Reported by: tzafrir ........ - -2009-09-08 14:28 +0000 [r216996] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | - dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines - caller id number empty parse_uri was not being given the correct - scheme's, as a result, uri parsing did not parse the username - correctly. One of the side effects of this is an empty caller id. - (closes issue #15839) Reported by: ebroad Patches: - blank_cidv2.patch uploaded by ebroad (license 878) - parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: - ebroad, dvossel ........ - -2009-09-07 16:38 +0000 [r216645-216843] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | - oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines - Make sure we reset global_exclude_static at channel reload - ........ - - * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | - oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If - there is no session timer in the INVITE, set it to default value - (not unset minimum = -1) Patch by oej closes issue #15621 - Reported by: fnordian Tested by: atis ........ - - * configs/sip.conf.sample: fix documentation so it agrees with code - - * channels/chan_sip.c, CHANGES: Add doc and turn off premature - media filter by default - - * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, - apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, - 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 - lines Make apps send PROGRESS control frame for early media and - fix too early media issue in SIP The issue at hand is that some - legacy (dying) PBX systems send empty media frames on PRI links - *before* any call progress. The SIP channel receives these frames - and by default signals 183 Session progress and starts sending - media. This will cause phones to play silence and ignore the - later 180 ringing message. A bad user experience. The fix is - twofold: - We discovered that asterisk apps that support early - media ("noanswer") did not send any PROGRESS frame to indicate - early media. Fixed. - We introduce a setting in chan_sip so that - users can disable any relay of media frames before the outbound - channel actually indicates any sort of call progress. In 1.4, - 1.6.0 and 1.6.1, this will be disabled for backward - compatibility. In later versions of Asterisk, this will be - enabled. We don't assume that it will change your Asterisk phone - experience - only for the better. We encourage third-party - application developers to make sure that if they have - applications that wants to send early media, add a PROGRESS - control frame transmission to make sure that all channel drivers - actually will start sending early media. This has not been the - default in Asterisk previous to this patch, so if you got - inspiration from our code, you need to update accordingly. Sorry - for the trouble and thanks for your support. This code has been - running for a few months in a large scale installation (over 250 - servers with PRI and/or BRI links to old PBX systems). That's no - proof that this is an excellent patch, but, well, it's tested :-) - ........ ................ - -2009-09-04 19:32 +0000 [r216595] Sean Bright <sean@malleable.com> - - * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep - 2009) | 1 line Use ast_free() instead of free(). ........ - -2009-09-04 17:32 +0000 [r216548] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 - Sep 2009) | 3 lines Enable turning off the application delimiter - warning with the 'dontwarn' option. Suggested on the -dev list, - and implemented in an alternate way by me. ........ - -2009-09-04 15:07 +0000 [r216507] Michiel van Baak <michiel@vanbaak.info> - - * /, main/utils.c: Merged revisions 216506 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) - | 9 lines Merged revisions 216435 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) - | 2 lines make asterisk compile under devmode with DEBUG_THREADS - enabled on OpenBSD ........ ................ - -2009-09-04 10:49 +0000 [r216265] Russell Bryant <russell@digium.com> - - * doc/IAX2-security.txt (added), /: Merged revisions 216264 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r216264 | russell | 2009-09-04 05:48:44 -0500 - (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r216263 | russell | 2009-09-04 05:48:00 -0500 - (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 - Sep 2009) | 2 lines Add a plain text version of the IAX2 security - document. ........ ................ ................ - -2009-09-04 06:11 +0000 [r216223] Michiel van Baak <michiel@vanbaak.info> - - * main/astobj2.c, /: Merged revisions 216222 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 | - mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines - make sure 'start' is always initialized. Makes asterisk compile - with --enable-dev-mode ........ - -2009-09-03 19:42 +0000 [r216011-216097] Russell Bryant <russell@digium.com> - - * UPGRADE.txt: tweak - - * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) - | 16 lines Merged revisions 216085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r216085 | russell | 2009-09-03 14:36:46 -0500 - (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 - Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. - ........ ................ ................ - - * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r216009 | russell | 2009-09-03 13:45:54 -0500 - (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r216008 | russell | 2009-09-03 13:44:58 -0500 - (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 - Sep 2009) | 2 lines Add IAX2 security document related to - AST-2009-006. ........ ................ ................ - -2009-09-03 18:40 +0000 [r216003] David Vossel <dvossel@digium.com> - - * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample, - include/asterisk/acl.h, channels/iax2-parser.h, /, - include/asterisk/astobj2.h, channels/iax2.h, main/acl.c, - channels/chan_iax2.c: Merged revisions 215955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | - dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines - Merge code associated with AST-2009-006 (closes issue #12912) - Reported by: rathaus Tested by: tilghman, russell, dvossel, - dbrooks ........ - -2009-09-03 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.0.15 released - - * AST-2009-006 - -2009-08-28 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.0.14 released - -2009-08-11 Tilghman Lesher <tlesher@digium.com> - - * Released 1.6.0.14-rc1 - -2009-08-10 19:51 +0000 [r211551-211587] Tilghman Lesher <tlesher@digium.com> - - * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 - (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 - Aug 2009) | 1 line Conversion specifiers, not format specifiers - ........ ................ - - * res/res_config_curl.c, apps/app_waitforring.c, - channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c, - pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c, - res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c, - doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c, - main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c, - apps/app_followme.c, channels/misdn_config.c, utils/frame.c, - main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c, - channels/chan_phone.c, apps/app_osplookup.c, - apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c, - apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c, - apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, - funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c, - res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c, - apps/app_waituntil.c, main/acl.c, apps/app_queue.c, - channels/chan_oss.c, agi/eagi-sphinx-test.c, - channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c, - apps/app_sms.c, utils/extconf.c, apps/app_verbose.c, - apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c, - apps/app_readfile.c, main/frame.c, /, apps/app_record.c, - funcs/func_strings.c, cdr/cdr_adaptive_odbc.c, - apps/app_alarmreceiver.c, channels/chan_iax2.c, - main/indications.c, main/config.c, main/cli.c, - pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c, - res/res_smdi.c, channels/chan_skinny.c, main/features.c, - main/http.c, main/pbx.c, apps/app_privacy.c, - codecs/codec_speex.c, funcs/func_math.c, channels/chan_agent.c, - apps/app_morsecode.c, apps/app_disa.c, funcs/func_cut.c, - channels/iax2-provision.c, pbx/dundi-parser.c, - apps/app_talkdetect.c: AST-2009-005 - -2009-08-10 14:10 +0000 [r211348] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | - file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix - retrieval of the port used for the video stream when adding SDP - to a SIP message. (closes issue #15121) Reported by: jsmith - ........ - -2009-08-09 15:43 +0000 [r211233-211276] Tilghman Lesher <tlesher@digium.com> - - * /, main/astfd.c: Merged revisions 211275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) - | 9 lines Merged revisions 211274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) - | 2 lines Small oops. Clear the flags which have been checked. - ........ ................ - - * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | - tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines - Check for NULL frame, before dereferencing pointer. (closes issue - #15617) Reported by: rain ........ - -2009-08-07 20:14 +0000 [r211114] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) - | 11 lines Recorded merge of revisions 211112 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) - | 4 lines Resolve a deadlock involving app_chanspy and - masquerades. (ABE-1936) ........ ................ - -2009-08-07 18:18 +0000 [r211044] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 211040 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) - | 21 lines Merged revisions 211038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) - | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, - not the membername. This is a partial revert of revision 82590, - which was an attempted cleanup, but in reality, it broke - QUEUE_MEMBER_LIST, which has always been intended as a method by - which component interfaces could be queried from the queue. - Membername isn't useful here, because that field cannot be used - to obtain further information about the member. See the - documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, - QUEUE_MEMBER_PENALTY, and the various AMI commands which take a - member argument for further justification. (closes issue #15664) - Reported by: rain Patches: app_queue-queue_member_list.diff - uploaded by rain (license 327) ........ ................ - -2009-08-07 13:08 +0000 [r210993] Kevin P. Fleming <kpfleming@digium.com> - - * main/udptl.c, /: Merged revisions 210992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | - kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 - lines Workaround broken T.38 endpoints that offer tiny - MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as - the maximum IFP size that should be sent to them, rather than the - maximum packet payload size. If such an endpoint also requests - UDPRedundancy as the error correction mode, we'll end up - calculating a tiny maximum IFP size, so small as to be unusable. - This patch sets a lower bound on what we'll consider the remote's - maximum IFP size to be, assuming that endpoints that do this - really can accept larger packets than they've offered to accept. - (closes issue #15649) Reported by: dazza76 ........ - -2009-08-06 21:46 +0000 [r210909-210915] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 210914 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) - | 14 lines Merged revisions 210913 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) - | 7 lines Because channel information can be accessed outside of - the channel thread, we must lock the channel prior to modifying - it. (closes issue #15397) Reported by: caspy Patches: - 20090714__issue15397.diff.txt uploaded by tilghman (license 14) - Tested by: caspy ........ ................ - - * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged - revisions 210908 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | - tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines - Allow Gosub to recognize quote delimiters without consuming them. - (closes issue #15557) Reported by: rain Patches: - 20090723__issue15557.diff.txt uploaded by tilghman (license 14) - Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ - ........ - -2009-08-06 17:47 +0000 [r210818] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | - file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines - Accept additional T.38 reinvites after an initial one has been - handled. Discussion of this subject has yielded that it is not - actually acceptable to change T.38 parameters after the initial - reinvite but declining is harsh and can cause the fax to fail - when it may be possible to allow it to continue. This patch - changes things so that additional T.38 reinvites are accepted but - parameter changes ignored. This gives the fax a fighting chance. - (closes issue #15610) Reported by: huangtx2009 ........ - -2009-08-05 20:07 +0000 [r210647] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 - (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) - | 14 lines Dialplan starts execution before the channel setup is - complete. * Issue 15655: For the case where dialing is complete - for an incoming call, dahdi_new() was asked to start the PBX and - then the code set more channel variables. If the dialplan hungup - before these channel variables got set, asterisk would likely - crash. * Fixed potential for overlap incoming call to erroneously - set channel variables as global dialplan variables if the - ast_channel structure failed to get allocated. * Added missing - set of CALLINGSUBADDR in the dialing is complete case. (closes - issue #15655) Reported by: alecdavis ........ ................ - -2009-08-05 18:57 +0000 [r210568] Leif Madsen <lmadsen@digium.com> - - * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 - (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) - | 11 lines Update imapstorage.txt documentation. Updated the - imapstorage.txt documentation to reflect that issues with - c-client versions older than 2007 seem to cause crashing issues - that are not seen with more recent versions. Documentation has - been updated to reflect this. (closes issue #14496) Reported by: - vbcrlfuser Patches: __20090727-imap-documentation-patch.txt - uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, - dbrooks ........ ................ - -2009-08-04 14:54 +0000 [r210239] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 210238 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug - 2009) | 16 lines Merged revisions 210237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug - 2009) | 10 lines Eliminate spurious compiler warnings from system - headers on *BSD platforms. Ensure that system headers located in - /usr/local/include are actually treated as system headers by the - compiler, and not as local headers which are subject to warnings - from the -Wundef compiler option and others. (closes issue - #15606) Reported by: mvanbaak ........ ................ - -2009-08-01 11:31 +0000 [r209840-209896] Russell Bryant <russell@digium.com> - - * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r209887 | russell | 2009-08-01 06:29:25 -0500 - (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) - | 5 lines Resolve a valgrind warning about a read from - uninitialized memory. (issue #15396) Reported by: aragon ........ - ................ - - * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r209839 | russell | 2009-08-01 06:02:07 -0500 - (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) - | 13 lines Modify how Playtones() is used in Milliwatt() to - resolve gain issue. When Milliwatt() was changed internally to - use Playtones() so that the proper tone was used, it introduced a - drop in gain in the output signal. So, use the playtones API - directly and specify a volume argument such that the output - matches the gain of the original Milliwatt() code. (closes issue - #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff - uploaded by russell (license 2) Tested by: rue_mohr ........ - ................ - -2009-08-01 01:13 +0000 [r209762] Kevin P. Fleming <kpfleming@digium.com> - - * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile, - channels/misdn/ie.c, main/event.c: Merged revisions 209760-209761 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 - (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul - 2009) | 7 lines Minor changes inspired by testing with latest - GCC. The latest GCC (what will become 4.5.x) has a few new - warnings, that in these cases found some either downright buggy - code, or at least seriously poorly designed code that could be - improved. ........ ................ r209761 | kpfleming | - 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert - accidental Makefile change. ................ - -2009-07-31 21:56 +0000 [r209712] Russell Bryant <russell@digium.com> - - * /, main/event.c: Merged revisions 209711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | - russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines - Fix some places where ast_event_type was used instead of - ast_event_ie_type. ........ - -2009-07-30 16:37 +0000 [r209555-209587] David Brooks <dbrooks@digium.com> - - * channels/chan_dahdi.c: Merged revisions 209554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | - dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines - Fixes numerous spelling errors. Patch submitted by alecdavis. - (closes issue #15595) Reported by: alecdavis ........ - - * include/asterisk/abstract_jb.h, - contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, - codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions - 209554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | - dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines - Fixes numerous spelling errors. Patch submitted by alecdavis. - (closes issue #15595) Reported by: alecdavis ........ - -2009-07-28 12:01 +0000 [r209394] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_fax.c: Correct error in backport of latest app_fax - fixes. - -2009-07-28 00:19 +0000 [r209325] Tilghman Lesher <tlesher@digium.com> - - * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) - | 9 lines Merged revisions 209315 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) - | 2 lines Publish French extra sounds ........ ................ - -2009-07-27 21:44 +0000 [r209259-209280] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | - kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 - lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE - messages about T.38 negotiation in debug level 1 messages, clean - up some looping logic, and correct an improper use of ast_free() - for freeing an ast_frame. ........ - - * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | - kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 - lines Make T.38 switchover in ReceiveFAX synchronous. In receive - mode, if the channel that ReceiveFAX is running on supports T.38, - we should *always* attempt to switch T.38, rather than listening - for an incoming CNG tone and only triggering on that. The channel - may be using a low-bitrate codec that distorts the CNG tone, the - sending FAX endpoint may not send CNG at all, or there could be a - variety of other reasons that we don't detect it, but in all - those cases if T.38 is available we certainly want to use it. - ........ - -2009-07-27 20:23 +0000 [r209221] David Brooks <dbrooks@digium.com> - - * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, - include/asterisk/module.h, main/features.c, res/res_agi.c, - res/res_jabber.c: Merged revisions 209098 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | - dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines - Fixing typos. Replaces "recieved" with "received" and "initilize" - with "initialize" (closes issue #15571) Reported by: alecdavis - ........ - -2009-07-27 20:16 +0000 [r209133-209198] Mark Michelson <mmichelson@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul - 2009) | 9 lines Honor channel's music class when using realtime - music on hold. (closes issue #15051) Reported by: alexh Patches: - 15051.patch uploaded by mmichelson (license 60) Tested by: alexh - ........ - - * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions - 209132 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul - 2009) | 24 lines Merged revisions 209131 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul - 2009) | 18 lines Allow for UDPTL to use only even-numbered ports - if desired. There are some VoIP providers out there that will not - accept SDP offers with odd numbered UDPTL ports. While it is my - personal opinion that these VoIP providers are misinterpreting - RFC 2327, it really is not a big deal to play along with their - silly little games. Of course, since restricting UDPTL ports to - only even numbers reduces the range of available ports by half, - so the option to use only even port numbers is off by default. A - user can enable the behavior by setting use_even_ports=yes in - udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: - 15182.patch uploaded by mmichelson (license 60) Tested by: - CGMChris ........ ................ - -2009-07-27 16:06 +0000 [r209061] David Brooks <dbrooks@digium.com> - - * res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" - with "received". From issue #15360, forgot to apply to trunk and - other branches. - -2009-07-27 15:39 +0000 [r209057] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 209056 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | - kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 - lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and - underscore-variants to sub-makes. During the recent Makefile - improvements I made, it seemed the 'make' was automatically - carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so - I removed the explict export of them. However, there are some - circumstances where make does this, and some where it does not, - so I've brought them back to ensure they are always exported. I - also removed an extraneous double setting of _ASTLDFLAGS on *BSD - platforms. ........ - -2009-07-27 01:21 +0000 [r208925] Jeff Peeler <jpeeler@digium.com> - - * /, main/translate.c, channels/chan_iax2.c: Merged revisions - 208924 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) - | 9 lines Merged revisions 208923 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) - | 2 lines Fix logic errors from 208746 ........ ................ - -2009-07-25 06:24 +0000 [r208752] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_skinny.c, main/translate.c, - channels/chan_iax2.c: Merged revisions 208749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) - | 13 lines Merged revisions 208746 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) - | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly - trivial changes, but I did not know of any other way to fix the - "dereferencing type-punned pointer will break strict-aliasing - rules" error without creating a tmp variable in chan_skinny. - ........ ................ - -2009-07-24 18:49 +0000 [r208594] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) - | 14 lines Merged revisions 208592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) - | 7 lines Do not log an ERROR if autoservice_stop() returns -1. - This does not indicate an error. A return of -1 just means that - the channel has been hung up. (reported in #asterisk-dev) - ........ ................ - -2009-07-24 18:31 +0000 [r208589] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul - 2009) | 16 lines Merged revisions 208587 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul - 2009) | 10 lines Only send a BYE when hanging up a channel that - is up. For cases where Asterisk sends an INVITE and receives a - non 2XX final response, Asterisk would follow the INVITE - transaction by immediately sending a BYE, which was unnecessary. - (closes issue #14575) Reported by: chris-mac ........ - ................ - -2009-07-24 15:04 +0000 [r208468-208549] Kevin P. Fleming <kpfleming@digium.com> - - * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: - Merged revisions 208548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | - kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 - lines Resolve a T.38 negotiation issue left over from the - udptl-updates merge. The udptl-updates branch that was merged - yesterday failed to properly send back T.38 SDP responses with - the correct error correction mode, if the incoming SDP from the - other end caused us to change error correction modes. This patch - corrects that situation. ........ - - * UPGRADE.txt: Use correct formatting for T.38 change note in - UPGRADE.txt - - * main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /, - channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, - include/asterisk/udptl.h, include/asterisk/frame.h: Merged - revisions 208464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | - kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 - lines Rework of T.38 negotiation and UDPTL API to address - interoperability problems Over the past couple of months, a - number of issues with Asterisk negotiating (and successfully - completing) T.38 sessions with various endpoints have been found. - This patch attempts to address many of them, primarily focused - around ensuring that the endpoints' MaxDatagram size is honored, - and in addition by ensuring that T.38 session parameter - negotiation is performed correctly according to the ITU T.38 - Recommendation. The major changes here are: 1) T.38 applications - in Asterisk (app_fax) only generate/receive IFP packets, they do - not ever work with UDPTL packets. As a result of this, they - cannot be allowed to generate packets that would overflow the - other endpoints' MaxDatagram size after the UDPTL stack adds any - error correction information. With this patch, the application is - told the maximum *IFP* size it can generate, based on a - calculation using the far end MaxDatagram size and the active - error correction mode on the T.38 session. The same is true for - sending *our* MaxDatagram size to the remote endpoint; it is - computed from the value that the application says it can accept - (for a single IFP packet) combined with the active error - correction mode. 2) All treatment of T.38 session parameters as - 'capabilities' in chan_sip has been removed; these parameters are - not at all like audio/video stream capabilities. There are strict - rules to follow for computing an answer to a T.38 offer, and - chan_sip now follows those rules, using the desired parameters - from the application (or channel) that wants to accept the T.38 - negotiation. 3) chan_sip now stores and forwards - ast_control_t38_parameters structures for tracking 'our' and - 'their' T.38 session parameters; this greatly simplifies - negotiation, especially for pass-through calls. 4) Since T.38 - negotiation without specifying parameters or receiving the final - negotiated parameters is not very worthwhile, the AST_CONTROL_T38 - control frame has been removed. A note has been added to - UPGRADE.txt about this removal, since any out-of-tree - applications that use it will no longer function properly until - they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: - https://reviewboard.asterisk.org/r/310/ ........ - -2009-07-23 19:35 +0000 [r208389] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul - 2009) | 24 lines Merged revisions 208386 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul - 2009) | 17 lines Fix a problem where a 491 response could be sent - out of dialog. This generalizes the fix for issue 13849. The - initial fix corrected the problem that Asterisk would reply with - a 491 if a reinvite were received from an endpoint and we had not - yet received an ACK from that endpoint for the initial INVITE it - had sent us. This expansion also allows Asterisk to appropriately - handle an INVITE with authorization credentials if Asterisk had - not received an ACK from the previous transaction in which - Asterisk had responded to an unauthorized INVITE with a 407. - (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch - uploaded by mmichelson (license 60) Tested by: klaus3000 ........ - ................ - -2009-07-23 19:23 +0000 [r208384] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500 - (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) - | 6 lines Only set the priindication setting when not performing - a reload (closes issue #14696) Reported by: fdecher ........ - ................ - -2009-07-23 16:30 +0000 [r208264-208316] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul - 2009) | 9 lines Merged revisions 208312 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul - 2009) | 3 lines Remove inaccurate XXX comment. ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul - 2009) | 15 lines Merged revisions 208262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul - 2009) | 8 lines Properly handle 183 responses which do not - contain an SDP. (closes issue #15442) Reported by: ffloimair - Patches: 15442.patch uploaded by mmichelson (license 60) Tested - by: tkarl, ffloimair ........ ................ - -2009-07-21 22:47 +0000 [r207947] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 - (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) - | 8 lines Force an error if a blank is passed to QUOTE (because - the documentation states the argument is not optional). This - change makes URIENCODE and QUOTE behave similarly, since the - documentation states that the argument is not optional, for both. - (closes issue #15439) Reported by: pkempgen Patches: - 20090706__issue15439.diff.txt uploaded by tilghman (license 14) - ........ ................ - -2009-07-21 20:27 +0000 [r207783-207860] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 - (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) - | 9 lines Wait for wink before dialing when using E&M wink - signaling There was already code for other signaling types in - dahdi_handle_event to handle dialing if a dial operation dial - string was present. Simply add SIG_EMWINK to the list. (closes - issue #14434) Reported by: araasch ........ ................ - - * channels/chan_dahdi.c: Revert r207636, this approach could - potentially block for an unacceptable amount of time. - -2009-07-21 14:30 +0000 [r207725] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 207723 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul - 2009) | 11 lines Merged revisions 207714 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul - 2009) | 5 lines Document default timeout for AMI originations. - AST-224 ........ ................ - -2009-07-21 13:39 +0000 [r207683] Kevin P. Fleming <kpfleming@digium.com> - - * funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, - Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /, - main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, - Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile: - Merged revisions 207680 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul - 2009) | 18 lines Merged revisions 207647 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul - 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are - honored. This commit changes the build system so that - user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to - the compiler/linker *after* all flags provided by the build - system itself, so that the user can effectively override the - build system's flags if desired. In addition, ASTCFLAGS and - ASTLDFLAGS can now be provided *either* in the environment before - running 'make', or as variable assignments on the 'make' command - line. As a result, the use of COPTS and LDOPTS is no longer - necessary, so they are no longer documented, but are still - supported so as not to break existing build systems that supply - them when building Asterisk. ........ ................ - -2009-07-21 04:38 +0000 [r207636] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: Wait for wink before dialing when using - E&M wink signaling This patch adds a new dahdi_wait function to - specifically wait for the wink event. If the wink is not - eventually received the channel is hung up. (closes issue #14434) - Reported by: araasch Patches: emwinkmod uploaded by araasch - (license 693) - -2009-07-20 19:55 +0000 [r207425] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul - 2009) | 39 lines Merged revisions 207423 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul - 2009) | 33 lines Answer video SDP offers properly when - videosupport is not enabled. Copied from Review board: In issue - 12434, the reporter describes a situation in which audio and - video is offered on the call, but because videosupport is - disabled in sip.conf, Asterisk gives no response at all to the - video offer. According to RFC 3264, all media offers should have - a corresponding answer. For offers we do not intend to actually - reply to with meaningful values, we should still reply with the - port for the media stream set to 0. In this patch, we take note - of what types of media have been offered and save the information - on the sip_pvt. The SDP in the response will take into account - whether media was offered. If we are not otherwise going to - answer a media offer, we will insert an appropriate m= line with - the port set to 0. It is important to note that this patch is - pretty much a bandage being applied to a broken bone. The patch - *only* helps for situations where video is offered but - videosupport is disabled and when udptl_pt is disabled but T.38 - is offered. Asterisk is not guaranteed to respond to every media - offer. Notable cases are when multiple streams of the same type - are offered. The 2 media stream limit is still present with this - patch, too. In trunk and the 1.6.X branches, things will be a bit - different since Asterisk also supports text in SDPs as well. - (closes issue #12434) Reported by: mnnojd Review: - https://reviewboard.asterisk.org/r/311 Review: - https://reviewboard.asterisk.org/r/313 ........ ................ - -2009-07-20 16:37 +0000 [r207362] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 207361 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) - | 16 lines Merged revisions 207360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) - | 9 lines Only do the chan->fdno check in ast_read() in a - developer build. I changed this check to only happen in a - dev-mode build. I also added a comment explaining what is going - on. I also made it so that detection of this situation does not - affect ast_read() operation. (closes issue #14723) Reported by: - seadweller ........ ................ - -2009-07-18 01:35 +0000 [r207286] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, - doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c, - configs/misdn.conf.sample: Merged revisions 145293,158010 from - https://origsvn.digium.com/svn/asterisk/branches/1.4 to make - merging easier. These changes are already on trunk. - ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 - (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c - channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk - to make merging easier later. ........ r145200 | rmudgett | - 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * - Miscellaneous formatting changes to make v1.4 and trunk more - merge compatible in the mISDN area. channels/chan_misdn.c * - Eliminated redundant code in cb_events() EVENT_SETUP ........ - r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) - | 9 lines improved helptext of misdn_set_opt. ........ r142181 | - rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line - Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 - 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines - channels/chan_misdn.c * Made bearer2str() use - allowed_bearers_array[] * Made use the causes.h defines instead - of hardcoded numbers. * Made use Asterisk presentation indicator - values if either of the mISDN presentation or screen options are - negative. * Updated the misdn_set_opt application option - descriptions. * Renamed the awkward Caller ID presentation - misdn_set_opt application option value not_screened to - restricted. Deprecated the not_screened option value. - channels/misdn/isdn_lib.c * Made use the causes.h defines instead - of hardcoded numbers. * Fixed some spelling errors and typos. * - Added all defined facility code strings to fac2str(). - channels/misdn/isdn_lib.h * Added doxygen comments to struct - misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen - comments to struct misdn_stack. channels/misdn_config.c - configs/misdn.conf.sample * Updated the mISDN presentation and - screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) - * Updated the misdn_set_opt application option descriptions. * - Fixed some spelling errors and typos. ................ r158010 | - rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines - Merged revision 157977 from - https://origsvn.digium.com/svn/asterisk/team/group/issue8824 - ........ Fixes JIRA ABE-1726 The dial extension could be empty if - you are using MISDN_KEYPAD to control ISDN provider features. - ................ - -2009-07-17 19:38 +0000 [r207097-207157] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 - (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) - | 7 lines Fix format specifier to print out an unsigned long - long. Yep, it's even ifdefed out code. But it made it to the RR - list... (closes issue #14726) Reported by: lmadsen ........ - ................ - - * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 - Jul 2009) | 2 lines Update some missing allowed options for - overlapdial ........ - -2009-07-17 17:53 +0000 [r206871-207032] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | - dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines - sip option flags handled incorrectly (closes issue #15376) - Reported by: Takehiko Ooshima Tested by: dvossel, - Takehiko_Ooshima ........ - - * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) - | 20 lines Merged revisions 206938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) - | 14 lines SIP incorrect From: header information when callpres - is prohib Some ITSP make use of the "Anonymous" display name to - detect a requirement to withhold caller id across the PSTN. This - does not work if the display name is "Unknown". (closes issue - #14465) Reported by: Nick_Lewis Patches: - chan_sip.c-callerpres.patch uploaded by Nick (license 657) - chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license - 671) Tested by: Nick_Lewis, dvossel ........ ................ - - * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 - (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) - | 6 lines error in iax.conf related IP-based access control - (closes issue #15518) Reported by: pkempgen ........ - ................ - - * /, main/callerid.c: Merged revisions 206868 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) - | 14 lines Merged revisions 206867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) - | 8 lines avoid segfault caused by user error If the CALLERPRES() - dialplan function is set to nothing, a segfault occurs. This is - user error to begin with, but I'd rather see a cli warning - message than have Asterisk crash on me. ........ ................ - -2009-07-16 16:52 +0000 [r206809] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 - (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) - | 6 lines Fix a memory leak. (closes issue #15517) Reported by: - adomjan Patches: func_realtime.c-ast_variable_destroy.diff - uploaded by adomjan (license 487) ........ ................ - -2009-07-15 22:06 +0000 [r206775] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | - dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines - Session timer were not activated if Supported header field in - INVITE had both "timer" and other options. (closes issue #15403) - Reported by: makoto Patches: sip-session-timer.patch uploaded by - makoto (license ........ - -2009-07-15 21:34 +0000 [r206762] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: - Merged revisions 206707 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) - | 33 lines Merged revisions 206706 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 - (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from - https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... - .......... Fixed chan_misdn crash because mISDNuser library is - not thread safe. With Asterisk the mISDNuser library is driven by - two threads concurrently: 1. - channels/misdn/isdn_lib.c::manager_event_handler() 2. - channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls - into the library are done concurrently and recursively from - isdn_lib.c. Both threads can fiddle with the master/child - layer3_proc_t lists. One thread may traverse the list when the - other interrupts it and then removes the list element which the - first thread was currently handling. This is exactly what caused - the crash. About 60 calls were needed to a Gigaset CX475 before - it occurred once. This patch adds locking when calling into the - mISDNuser library. This also fixes some cb_log calls with wrong - port parameter. JIRA ABE-1913 Patches: misdn-locking.patch - (Modified with mostly cosmetic changes) .......... - ................ ................ - -2009-07-15 20:21 +0000 [r206705] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | - dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines - callerid(num) is wrong when username is missing A domain only sip - uri <sip:123.123.123.123> would return 123.123.123.123 as callid - num. Now, if the username is missing from a uri, the callerid num - field is left empty. (closes issue #15476) Reported by: viraptor - ........ - -2009-07-15 16:02 +0000 [r206637] Sean Bright <sean@malleable.com> - - * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 - (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, - 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we - are asking for it. ........ ................ - -2009-07-14 20:22 +0000 [r206585] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 - Jul 2009) | 6 lines Document all meetme realtime fields, and in - the process, make some field lengths more consistent. (closes - issue #15493) Reported by: lasko Patches: meetme.diff uploaded by - lasko (license 833) ........ - -2009-07-14 18:17 +0000 [r206555] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 - (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) - | 28 lines Fixes several call transfer issues with chan_misdn. * - issue #14355 - Crash if attempt to transfer a call to an - application. Masquerade the other pair of the four asterisk - channels involved in the two calls. The held call already must be - a bridged call (not an applicaton) or it would have been - rejected. * issue #14692 - Held calls are not automatically - cleared after transfer. Allow the core to initate disconnect of - held calls to the ISDN port. This also fixes a similar case where - the party on hold hangs up before being transferred or taken off - hold. * JIRA ABE-1903 - Orphaned held calls left in - music-on-hold. Do not simply block passing the hangup event on - held calls to asterisk core. * Fixed to allow held calls to be - transferred to ringing calls. Previously, held calls could only - be transferred to connected calls. * Eliminated unused call - states to simplify hangup code. * Eliminated most uses of - "holded" because it is not a word. (closes issue #14355) (closes - issue #14692) Reported by: sodom Patches: - misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) - Tested by: rmudgett ........ ................ - -2009-07-14 14:54 +0000 [r206387] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 206386 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r206386 | russell | 2009-07-14 09:51:44 -0500 - (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r206385 | russell | 2009-07-14 09:48:00 -0500 - (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) - | 6 lines Ensure apathetic replies are sent out on the proper - socket. chan_iax2 supports multiple address bindings. The - send_apathetic_reply() function did not attempt to send its - response on the same socket that the incoming message came in on. - ........ ................ ................ - -2009-07-14 01:25 +0000 [r206369] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 206341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) - | 11 lines Merged revisions 206284 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) - | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 - ........ ................ - -2009-07-10 22:50 +0000 [r206017] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | - dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines - SIP register not using peer's outbound proxy If callbackextension - is defined for a peer it successfully causes a registration to - occur, but the registration ignores the outboundproxy settings - for the peer. This patch allows the peer to be passed to - obproxy_get() in transmit_register(). (closes issue #14344) - Reported by: Nick_Lewis Patches: - callbackextension_peer_trunk.diff uploaded by dvossel (license - 671) Tested by: dvossel Review: - https://reviewboard.asterisk.org/r/294/ ........ - -2009-07-10 18:44 +0000 [r205940] Kevin P. Fleming <kpfleming@digium.com> - - * main/udptl.c, /: Merged revisions 205939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | - kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line - Update comments about the level of T.38 support in Asterisk. - ........ - -2009-07-10 17:44 +0000 [r205879-205880] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix build. - - * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul - 2009) | 30 lines Merged revisions 205877 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 - (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 - (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul - 2009) | 10 lines Ensure that outbound NOTIFY requests are - properly routed through stateful proxies. With this change, we - make note of Record-Route headers present in any SUBSCRIBE - request that we receive so that our outbound NOTIFY requests will - have the proper Route headers in them. (closes issue #14725) - Reported by: ibc ........ ................ ................ - ................ - -2009-07-10 16:48 +0000 [r205843] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) - | 37 lines Merged revisions 205804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) - | 31 lines SIP registration auth loop caused by stale nonce If an - endpoint sends two registration requests in a very short period - of time with the same nonce, both receive 401 responses from - Asterisk, each with a different nonce (the second 401 containing - the current nonce and the first one being stale). If the endpoint - responds to the first 401, it does not match the current nonce so - Asterisk sends a third 401 with a newly generated nonce (which - updates the current nonce)... Now if the endpoint responds to the - second 401, it does not match the current nonce either and - Asterisk sends a fourth 401 with a newly generated nonce... This - loop goes on and on. There appears to be a simple fix for this. - If the nonce from the request does not match our nonce, but is a - good response to a previous nonce, instead of sending a 401 with - a newly generated nonce, use the current one instead. This breaks - the loop as the nonce is not updated until a response is - received. Additional logic has been added to make sure no nonce - can be responded to twice though. (closes issue #15102) Reported - by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license - 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: - Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ - ................ - -2009-07-10 15:57 +0000 [r205777] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul - 2009) | 16 lines Merged revisions 205775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul - 2009) | 10 lines Ensure that outbound NOTIFY requests are - properly routed through stateful proxies. With this change, we - make note of Record-Route headers present in any SUBSCRIBE - request that we receive so that our outbound NOTIFY requests will - have the proper Route headers in them. (closes issue #14725) - Reported by: ibc ........ ................ - -2009-07-10 15:35 +0000 [r205771] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | - kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 - lines Fix some remaining T.38 negotiation problems in app_fax. - Revision 205696 did not quite fix all the issues with the T.38 - negotiation changes and app_fax; this patch corrects them, along - with a couple of other minor issues. (closes issue #15480) - Reported by: dimas Patches: test2-15480.patch uploaded by dimas - (license 88) ........ - -2009-07-09 23:46 +0000 [r205729] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) - | 21 lines No audio on calls from Asterisk to various ISDN - devices until DTMF sent by caller. Add missing clearing of the - dialing flag when the ISDN call is CONNECTED. (i.e. When libpri - generates the event PRI_EVENT_ANSWER.) (closes issue #15420) - Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt - uploaded by alecdavis (license 585) Tested by: scottbmilne, - alecdavis (closes issue #15416) Reported by: avinoash (closes - issue #15389) Reported by: alecdavis This patch should also fix - the following issue: (issue #15205) Reported by: vinsik ........ - -2009-07-09 21:26 +0000 [r205697] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h: - Merged revisions 205696 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | - kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 - lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 - switchover. Recent changes in T.38 negotiation in Asterisk caused - these applications to not respond when the other endpoint - initiated a switchover to T.38; this resulted in the T.38 - switchover failing, and the FAX attempt to be made using an audio - connection, instead of T.38 (which would usually cause the FAX to - fail completely). This patch corrects this problem, and the - applications will now correctly respond to the T.38 switchover - request. In addition, the response will include the appopriate - T.38 session parameters based on what the other end offered and - what our end is capable of. (closes issue #14849) Reported by: - afosorio ........ - -2009-07-09 16:21 +0000 [r205597-205608] David Vossel <dvossel@digium.com> - - * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 - (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 - Jul 2009) | 2 lines Changing ast_samp2tv to not use floating - point. ........ ................ - - * main/rtp.c, /, channels/chan_iax2.c, include/asterisk/frame.h: - Merged revisions 205479 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) - | 16 lines Merged revisions 205471 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) - | 10 lines Fixes 8khz assumptions Many calculations assume 8khz - is the codec rate. This is not always the case. This patch only - addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there - are other areas that make this assumption as well. Review: - https://reviewboard.asterisk.org/r/306/ ........ ................ - -2009-07-09 08:32 +0000 [r205533] Michiel van Baak <michiel@vanbaak.info> - - * /, main/ssl.c: Merged revisions 205532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | - mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines - pthread_self returns a pthread_t which is not an unsigned int on - all pthread implementations. Casting it to an unsigned int fixes - compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit - ........ - -2009-07-08 22:17 +0000 [r205415] David Vossel <dvossel@digium.com> - - * include/asterisk/devicestate.h, main/pbx.c, /, - main/devicestate.c, include/asterisk/pbx.h: Merged revisions - 205412 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) - | 12 lines Merged revisions 205409 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) - | 6 lines moving ast_devstate_to_extenstate to pbx.c from - devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This - change fixes a compile time error with chan_vpb as well. ........ - ................ - -2009-07-08 19:27 +0000 [r205351] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 205350 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul - 2009) | 20 lines Merged revisions 205349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul - 2009) | 14 lines Prevent phantom calls to queue members. If a - caller were to hang up while a periodic announcement or position - were being said, the return value for those functions would - incorrectly indicate that the caller was still in the queue. With - these changes, the problem does not occur. (closes issue #14631) - Reported by: latinsud Patches: queue_announce_ghost_call2.diff - uploaded by latinsud (license 745) (with small modification from - me) ........ ................ - -2009-07-08 18:20 +0000 [r205296] Jason Parker <jparker@digium.com> - - * config.guess, config.sub, /: Merged revisions 205291 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 - (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul - 2009) | 1 line Update config.guess and config.sub from the - savannah.gnu.org git repo. ........ ................ - -2009-07-08 17:01 +0000 [r205224] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: oops, fixing build - -2009-07-08 16:56 +0000 [r205220] David Vossel <dvossel@digium.com> - - * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 - (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) - | 10 lines ast_samp2tv needs floating point for 16khz audio In - ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The - .5 is currently stripped off because we don't calculate using - floating points. This causes madness with 16khz audio. (issue - ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ - ........ ................ - -2009-07-08 16:28 +0000 [r205200] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c: Merged revisions 205196 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) - | 9 lines Merged revisions 205188 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) - | 2 lines Add redirection warnings for the invalid language codes - previously removed. ........ ................ - -2009-07-08 15:56 +0000 [r205139-205152] Russell Bryant <russell@digium.com> - - * /, main/ssl.c: Merged revisions 205151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | - russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines - Use tabs instead of spaces for indentation. ........ - - * main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c - (added), include/asterisk/_private.h, res/res_jabber.c: Merged - revisions 205120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | - russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines - Move OpenSSL initialization to a single place, make library usage - thread-safe. While doing some reading about OpenSSL, I noticed a - couple of things that needed to be improved with our usage of - OpenSSL. 1) We had initialization of the library done in multiple - modules. This has now been moved to a core function that gets - executed during Asterisk startup. We already link OpenSSL into - the core for TCP/TLS functionality, so this was the most logical - place to do it. 2) OpenSSL is not thread-safe by default. - However, making it thread safe is very easy. We just have to - provide a couple of callbacks. One callback returns a thread ID. - The other handles locking. For more information, start with the - "Is OpenSSL thread-safe?" question on the FAQ page of - openssl.org. ........ - -2009-07-08 14:35 +0000 [r205117] David Vossel <dvossel@digium.com> - - * channels/chan_sip.c, include/asterisk/sched.h: SIP Dialog ref - counting This patch adds reference counting for sip dialogs into - 1.6.0. When proc_session_timer() is called from the scheduler - thread it has no guarantee the session timer's dialog won't be - freed from underneath it. Now the session timer holds a reference - to the dialog, preventing it from being destroyed during the - middle of proc_session_timer(). (closes issue #13623) Reported - by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ - -2009-07-06 15:17 +0000 [r204980] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: Restore Hungarian (mistakenly removed during merge) - -2009-07-06 13:39 +0000 [r204949] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, /: Merged revisions 204948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | - kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 - lines Improve handling of AST_CONTROL_T38 and - AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This - change allows applications that request T.38 negotiation on a - channel that does not support it to get the proper indication - that it is not supported, rather than thinking that negotiation - was started when it was not. ........ - -2009-07-02 22:03 +0000 [r204836] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 - (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) - | 10 lines Removed confusing warning message "Got Busy in - Connected State" If an incoming mISDN call is answered with the - Answer application and a subsequent Dial gets a busy endpoint - then it is valid for that already connected channel to get the - busy indication. Asterisk will play the busy tones until the - dialplan plays something else or hangs up the call. (closes issue - #11974) Reported by: fvdb ........ ................ - -2009-07-02 18:07 +0000 [r204652-204754] David Vossel <dvossel@digium.com> - - * include/asterisk/devicestate.h, main/pbx.c, /, - main/devicestate.c: Merged revisions 204710 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) - | 21 lines Merged revisions 204681 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) - | 14 lines Improved mapping of extension states from combined - device states. This fixes a few issues with incorrect extension - states and adds a cli command, core show device2extenstate, to - display all possible state mappings. (closes issue #15413) - Reported by: legart Patches: exten_helper.diff uploaded by - dvossel (license 671) Tested by: dvossel, legart, amilcar Review: - https://reviewboard.asterisk.org/r/301/ ........ ................ - - * channels/chan_sip.c: removes fake dialog_unref and dialog_ref - function calls. dialog_unref() and dialog_ref() in 1.6.0 where - only place holders for reference counting once it was - implemented. The functions did nothing but return the pointer on - ref and NULL on unref. These calls have been removed to make way - for a patch that actually does dialog ref counting. - -2009-06-30 21:21 +0000 [r204581] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 - (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) - | 6 lines More incorrect language codes, plus ensuring that - regionalizations use the specified language, and not English for - grammar. (closes issue #15022) Reported by: greenfieldtech - Patches: 20090519__issue15022.diff.txt uploaded by tilghman - (license 14) ........ ................ - -2009-06-30 18:50 +0000 [r204476] Jason Parker <jparker@digium.com> - - * /, main/say.c: Merged revisions 204475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | - 9 lines Merged revisions 204474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | - 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a - comment typo in passing. ........ ................ - -2009-06-30 18:44 +0000 [r204471] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge - of revisions 204470 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) - | 18 lines Recorded merge of revisions 204469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) - | 11 lines "tw" is the language specification for Twi (from - Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier - Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman - (license 14) 20090617__issue15346__trunk.diff.txt uploaded by - tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt - uploaded by tilghman (license 14) - 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman - (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by - tilghman (license 14) Tested by: volivier ........ - ................ - -2009-06-29 22:52 +0000 [r204248-204302] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun - 2009) | 15 lines Merged revisions 204300 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun - 2009) | 9 lines Add error message so that it is clear why a SIP - peer was not processed when a DNS lookup fails on a host or - outboundproxy. (closes issue #13432) Reported by: p_lindheimer - Patches: outboundproxy.patch uploaded by p (license 558) ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun - 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun - 2009) | 22 lines Fix a problem where chan_sip would ignore "old" - but valid responses. chan_sip has had a problem for quite a long - time that would manifest when Asterisk would send multiple SIP - responses on the same dialog before receiving a response. The - problem occurred because chan_sip only kept track of the highest - outgoing sequence number used on the dialog. If Asterisk sent two - requests out, and a response arrived for the first request sent, - then Asterisk would ignore the response. The result was that - Asterisk would continue retransmitting the requests and ignoring - the responses until the maximum number of retransmissions had - been reached. The fix here is to rearrange the code a bit so that - instead of simply comparing the sequence number of the response - to our latest outgoing sequence number, we walk our list of - outstanding packets and determine if there is a match. If there - is, we continue. If not, then we ignore the response. In doing - this, I found a few completely useless variables that I have now - removed. (closes issue #11231) Reported by: flefoll Review: - https://reviewboard.asterisk.org/r/298 ........ r204246 | - mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 - lines Fix build oops. ........ ................ - -2009-06-27 01:14 +0000 [r203910] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 - (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) - | 16 lines The ISDN CPE side should not exclusively pick B - channels normally. Before this patch, Asterisk unconditionally - picked B channels exclusively on the CPE side and normally - allowed alternative B channels on the network side. Now Asterisk - does the opposite. Reasons for the CPE side to normally not pick - B channels exclusively: * For CPE point-to-multipoint mode (i.e. - phone side), the CPE side does not have enough information to - exclusively pick B channels. (There may be other devices on the - line.) * Q.931 gives preference to the network side picking B - channels. * Some telcos require the CPE side to not pick B - channels exclusively. (closes issue #14383) Reported by: - mbrancaleoni ........ ................ - -2009-06-26 22:12 +0000 [r203855] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 - (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) - | 5 lines Make sure to recreate the dahdi pseudo channel after - dahdi restart (closes issue #14477) Reported by: timking ........ - ................ - -2009-06-26 21:25 +0000 [r203780-203818] Russell Bryant <russell@digium.com> - - * /, main/file.c: Merged revisions 203802 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) - | 22 lines Merged revisions 203785 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) - | 15 lines Don't fast forward past the end of a message. This is - nice change for users of the voicemail application. If someone - gets a little carried away with fast forwarding through a - message, they can easily get to the end and accidentally exit the - voicemail application by hitting the fast forward key during the - following prompt. This adds some safety by not allowing a fast - forward past the end of a message. (closes issue #14554) Reported - by: lacoursj Patches: 21761.patch uploaded by lacoursj (license - 707) Tested by: lacoursj ........ ................ - - * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | - russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines - Ensure the TCP read buffer is fully initialized before handling - each packet. (closes issue #14452) Reported by: umberto71 - ........ - -2009-06-26 20:16 +0000 [r203722] David Brooks <dbrooks@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) - | 16 lines Fixing voicemail's error in checking max silence vs - min message length Max silence was represented in milliseconds, - yet vmminsecs (minmessage) was represented as seconds. Also, the - inequality was reversed. The warning, if triggered, was "Max - silence should be less than minmessage or you may get empty - messages", which should have been logged if max silence was - greater than minmessage, but the check was for less than. Also, - conforming if statement to coding guidelines. closes issue - #15331) Reported by: markd Review: - https://reviewboard.asterisk.org/r/293/ ........ - -2009-06-26 19:54 +0000 [r203717] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: reverse whitespace change 203711 that was - based on looking at sig_analog (which has about a 1000 line - indentation change that is not worth doing here) - -2009-06-26 19:49 +0000 [r203716] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 203710 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) - | 7 lines moving debug message from level 0 to 1. (closes issue - #15404) Reported by: leobrown Patches: iax_codec_debug.patch - uploaded by leobrown (license 541) ........ - -2009-06-26 19:47 +0000 [r203711] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: whitespace fix - -2009-06-26 19:29 +0000 [r203701] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, - apps/app_fax.c, configs/sip.conf.sample, - include/asterisk/frame.h: Merged revisions 203699 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 - lines Improve T.38 negotiation by exchanging session parameters - between application and channel. ........ - -2009-06-26 19:25 +0000 [r203698] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) - | 16 lines Check if polarityonanswerdelay has elapsed before - setting a channel as answered after a polarity reversal. - Previously on a polarity switch event chan_dahdi would set the - channel immediately as answered. This would cause problems if a - polarity reversal occurred when the line was picked up as the - dial would not have yet occurred. Now if the polarity reversal - occurs before delay has elapsed after coming off hook or an - answer, it is ignored. Also, some refactoring was done in - _handle_event. (closes issue #13917) Reported by: alecdavis - Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by - alecdavis (license 585) Tested by: alecdavis ........ - -2009-06-25 21:47 +0000 [r203447] David Vossel <dvossel@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 - Jun 2009) | 4 lines fixes a few redundant conditions (issue - #15269) ........ - -2009-06-25 21:18 +0000 [r203387] Terry Wilson <twilson@digium.com> - - * main/cli.c, /: Merged revisions 203381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) - | 11 lines Merged revisions 203380 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) - | 4 lines I didn't see that Mark already fixed the underlying - issue! Yay for removing useless code. ........ ................ - -2009-06-25 21:06 +0000 [r203117-203377] Russell Bryant <russell@digium.com> - - * /, main/features.c: Merged revisions 203376 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) - | 16 lines Merged revisions 203375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) - | 9 lines Fix a case where CDR answer time could be before the - start time involving parking. (closes issue #13794) Reported by: - davidw Patches: 13794.patch uploaded by murf (license 17) - 13794.patch.160 uploaded by murf (license 17) Tested by: murf, - dbrooks ........ ................ - - * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) - | 18 lines Merged revisions 203115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) - | 11 lines Resolve a crash related to a T.38 reinvite race - condition. This change resolves a crash observed locally during - some T.38 testing. A call was set up using a call file, and when - the T.38 reinvite came in, the channel state was still - AST_STATE_DOWN. The reason is explained by a comment in the code - that previously lived in the handling of AST_STATE_RINGING. This - change modifies the logic to handle the same race condition for - any channel state that is not UP. (closes ABE-1895) ........ - ................ - -2009-06-24 21:18 +0000 [r203044] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 - (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) - | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid - format is: pritimer=timer_name,timer_value * Fixed segfault if - the ',' is missing. * Completely check the range returned by - pri_timer2idx() to prevent possible access outside array bounds. - ........ ................ - -2009-06-24 18:29 +0000 [r202968] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun - 2009) | 9 lines Merged revisions 202966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun - 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding - the same thing in-line. ........ ................ - -2009-06-24 18:09 +0000 [r202926] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | - file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines - Ensure the default settings are applied for T.38 when we set it - up for a peer. ........ - -2009-06-23 22:09 +0000 [r202763] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | - 1 line I could have sworn I committed this patch ages ago, but... - bug fix with setting NAI properly on linksets in certain - situations. ........ - -2009-06-23 21:26 +0000 [r202754] Ryan Brindley <rbrindley@digium.com> - - * main/config.c, /: Merged revisions 202753 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | - rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 - lines If we delete the info, lets also delete the lines (closes - issue #14509) Reported by: timeshell Patches: - 20090504__bug14509.diff.txt uploaded by tilghman (license 14) - Tested by: awk, timeshell ........ - -2009-06-23 16:40 +0000 [r202675] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) - | 18 lines Merged revisions 202671 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) - | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to - non-standard port and transport (closes issue #14659) Reported - by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded - by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded - by dvossel (license 671) Tested by: dvossel, klaus3000 Review: - https://reviewboard.asterisk.org/r/288/ ........ ................ - -2009-06-22 20:12 +0000 [r202498] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 202497 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) - | 11 lines Merged revisions 202496 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) - | 4 lines Report CallerID change during a masquerade. Reported - by: markster ........ ................ - -2009-06-22 16:30 +0000 [r202471] Sean Bright <sean@malleable.com> - - * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun - 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid - potential crashes during reload. Pointed out by Russell while - working on the CEL branch. ........ - -2009-06-22 16:06 +0000 [r202416] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) - | 9 lines Merged revisions 202414 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) - | 2 lines Make Polycom subscription type override check more - explicit. ........ ................ - -2009-06-22 15:05 +0000 [r202338-202344] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun - 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun - 2009) | 26 lines Fix a situation in which Asterisk would not stop - retransmitting 487s. If a CANCEL were received by Asterisk, we - would send a 487 in response to the original INVITE and a 200 OK - for the CANCEL. If there were a network hiccup which caused the - 200 OK and the 487 to be lost, then the UA communicating with - Asterisk may try to retransmit its CANCEL. Asterisk's response to - this used to be to try sending another 487 to the canceled INVITE - and another 200 OK to the CANCEL. The problem here is that the - originally-sent 487 was sent "reliably" meaning that it will be - retransmitted until it is received properly. So when we receive - the second CANCEL it is likely that the first batch of 487s we - sent is still going strong and reaches the UA. The result was - that the second set of 487s would be retransmitted constantly - until the maximum number of retries had been reached. The fix for - this is that if we receive a second CANCEL for an INVITE, then we - cancel the retransmission of the first set of 487s and start a - second set. This causes the dialog to be terminated reasonably. - (closes issue #14584) Reported by: klaus3000 Patches: - 14584_v2.patch uploaded by mmichelson (license 60) Tested by: - klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 - -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line - left from previous commit. ........ ................ - - * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun - 2009) | 31 lines Merged revisions 202336 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun - 2009) | 25 lines Fix a possible infinite loop in SDP parsing - during glare situation. There was a while loop in - get_ip_and_port_from_sdp which was controlled by a call to - get_sdp_iterate. The loop would exit either if what we were - searching for was found or if the return was NULL. The problem is - that get_sdp_iterate never returns NULL. This means that if what - we were searching for was not present, the loop would run - infinitely. This modification of the loop fixes the problem. - (closes issue #15213) Reported by: schmidts (closes issue #15349) - Reported by: samy (closes issue #14464) Reported by: pj (closes - issue #15345) Reported by: aragon Patches: sip_inf_loop.patch - uploaded by mmichelson (license 60) Tested by: aragon ........ - ................ - -2009-06-21 16:14 +0000 [r202259-202263] Russell Bryant <russell@digium.com> - - * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | - russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines - Fix possibility of crashiness during reload in custom fields - handling. ........ - - * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | - russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines - Standardize return values of load_config() so reload() doesn't - report an error on success. ........ - -2009-06-20 19:14 +0000 [r202184] Sean Bright <sean@malleable.com> - - * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | - seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 - lines Fix version detection for API changes in spandsp. (closes - issue #15355) Reported by: deuffy ........ - -2009-06-19 21:07 +0000 [r202006] Matthew Nicholson <mnicholson@digium.com> - - * channels/chan_sip.c: Added deadlock protection to - try_suggested_sip_codec in chan_sip.c. Review: - https://reviewboard.asterisk.org/r/287/ - -2009-06-19 20:27 +0000 [r201997] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 201994 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 - (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) - | 8 lines timestamp was being converted to host order as a short - rather than a long (closes issue #15361) Reported by: ffloimair - Patches: ts_issue.diff uploaded by dvossel (license 671) ........ - ................ - -2009-06-19 00:44 +0000 [r201786-201830] Tilghman Lesher <tlesher@digium.com> - - * /, main/features.c: Merged revisions 201829 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) - | 13 lines Merged revisions 201828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) - | 6 lines If the "h" extension fails, give it another chance in - main/pbx.c. If the "h" extension fails, give it another chance in - main/pbx.c, when it returns from the bridge code. Fixes an issue - where the "h" extension may occasionally not fire, when a Dial is - executed from a Macro. Debugged in #asterisk with user tompaw. - ........ ................ - - * /, apps/Makefile: Merged revisions 201783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | - tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines - One of the changes in 1.6.1 was to allow app_directory to use - functionality within app_voicemail for directory functions. It is - therefore no longer necessary for app_directory to be linked - against the ODBC libraries (and it never was necessary for - app_directory to be linked against IMAP, though it was). ........ - -2009-06-18 16:58 +0000 [r201682] David Vossel <dvossel@digium.com> - - * channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c, - main/ast_expr2.c, utils/stereorize.c, - codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx, - main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, - utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: Merged - revisions 201678 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | - dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines - fixes some memory leaks and redundant conditions (closes issue - #15269) Reported by: contactmayankjain Patches: patch.txt - uploaded by contactmayankjain (license 740) - memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) - Tested by: contactmayankjain, dvossel ........ - -2009-06-18 15:32 +0000 [r201612] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r201610 | russell | 2009-06-18 10:27:10 -0500 - (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) - | 29 lines Fix memory corruption and leakage related reloads of - non files mode MoH classes. For Music on Hold classes that are - not files mode, meaning that we are executing an application that - will feed us audio data, we use a thread to monitor the external - application and read audio from it. This thread also makes use of - the MoH class object. In the MoH class destructor, we used - pthread_cancel() to ask the thread to exit. Unfortunately, the - code did not wait to ensure that the thread actually went away. - What needed to be done is a pthread_join() to ensure that the - thread fully cleans up before we proceed. By adding this one - line, we resolve two significant problems: 1) Since the thread - was never joined, it never fully goes away. So, on every reload - of non-files mode MoH, an unused thread was sticking around. 2) - There was a race condition here where the application monitoring - thread could still try to access the MoH class, even though the - thread executing the MoH reload has already destroyed it. (issue - #15109) Reported by: jvandal (issue #15123) Reported by: - axisinternet (issue #15195) Reported by: amorsen (issue AST-208) - ........ ................ - -2009-06-17 20:10 +0000 [r201459-201463] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | - mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 - lines Fix problem with no audio due to ignoring the SDP. A recent - change to our SDP version comparison made audio not function on - some calls. This was because of a test wherein we were trying to - see if an unsigned value was less than 0. This is a dumb - comparison and arguably the compiler should have warned about it. - Alas, though, it slipped past. Now it's fixed by changing the - variable to be a signed type. Found by several developers. Tested - by mnicholson and dbrooks. ........ - - * main/channel.c, /: Merged revisions 201458 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun - 2009) | 15 lines Merged revisions 201450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun - 2009) | 9 lines Change the datastore traversal in - ast_do_masquerade to use a safe list traversal. It is possible - for datastore fixup functions to remove the datastore from the - list and free it. In particular, the queue_transfer_fixup in - app_queue does this. While I don't yet know of this causing any - crashes, it certainly could. Found while discussing a separate - issue with Brian Degenhardt. ........ ................ - -2009-06-17 19:55 +0000 [r201449] David Vossel <dvossel@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 - (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) - | 19 lines StopMixMonitor race condition (not giving up file - immediately) StopMixMonitor only indicates to the MixMonitor - thread to stop writing to the file. It does not guarantee that - the recording's file handle is available to the dialplan - immediately after execution. This results in a race condition. To - resolve this, the filestream pointer is placed in a datastore on - the channel. When StopMixMonitor is called, the datastore is - retrieved from the channel and the filestream is closed - immediately before returning to the dialplan. Documentation - indicating the use of StopMixMonitor to free files has been - updated as well. (closes issue #15259) Reported by: travisghansen - Tested by: dvossel Review: - https://reviewboard.asterisk.org/r/283/ ........ ................ - -2009-06-17 19:35 +0000 [r201443] David Brooks <dbrooks@digium.com> - - * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) - | 16 lines Merged revisions 201380 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) - | 9 lines Checks for NULL sip_pvt pointer in - chan_sip.c->acf_channel_read() Zombie channels could be passed, - and chan_sip.c wasn't checking for it. Could crash Asterisk. Now - checking for NULL pointer. (closes issue #15330) Reported by: - okrief Tested by: dbrooks ........ ................ - -2009-06-17 12:05 +0000 [r201263] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 201262 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 - (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun - 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list - to be appended is empty. When the list to be appended is empty, - and the list to be appended to is *not*, AST_LIST_APPEND_LIST - would actually cause the target list to become broken, and no - longer have a pointer to its last entry. This patch fixes the - problem. (reported by Stanislaw Pitucha on the asterisk-dev - mailing list) ........ ................ - -2009-06-16 22:31 +0000 [r201226] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | - dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines - fix issue with build_contact introduced by the "SIP trasnport - type issues" commit ........ - -2009-06-16 19:34 +0000 [r201093] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, - main/autoservice.c, main/frame.c, /, apps/app_meetme.c, - configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4, - configure.ac, include/asterisk/linkedlists.h, main/file.c, - include/asterisk/channel.h, include/asterisk/frame.h: Merged - revisions 201056,201090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun - 2009) | 18 lines Merged revisions 200991 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun - 2009) | 11 lines Improve support for media paths that can - generate multiple frames at once. There are various media paths - in Asterisk (codec translators and UDPTL, primarily) that can - generate more than one frame to be generated when the application - calling them expects only a single frame. This patch addresses a - number of those cases, at least the primary ones to solve the - known problems. In addition it removes the broken TRACE_FRAMES - support, fixes a number of bugs in various frame-related API - functions, and cleans up various code paths affected by these - changes. https://reviewboard.asterisk.org/r/175/ ........ - ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 - (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler - attribute checking. Defaulting to 'static' for the function scope - was bad... so remove it. ................ - -2009-06-16 17:11 +0000 [r200992] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | - dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines - SIP transport type issues What this patch addresses: 1. - ast_sip_ouraddrfor() by default binds to the UDP address/port - reguardless if the sip->pvt is of type UDP or not. Now when no - remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's - transport type, attempting to set the address and port to the - correct TCP/TLS bindings if necessary. 2. It is not necessary to - send the port number in the Contact header unless the port is - non-standard for the transport type. This patch fixes this and - removes the todo note. 3. In sip_alloc(), the default dialog - built always uses transport type UDP. Now sip_alloc() looks at - the sip_request (if present) and determines what transport type - to use by default. 4. When changing the transport type of a - sip_socket, the file descriptor must be set to -1 and in some - cases the tcptls_session's ref count must be decremented and set - to NULL. I've encountered several issues associated with this - process and have created a function, set_socket_transport(), to - handle the setting of the socket type. (closes issue #13865) - Reported by: st Patches: dont_add_port_if_tls.patch uploaded by - Kristijan (license 753) 13865.patch uploaded by mmichelson - (license 60) tls_port_v5.patch uploaded by vrban (license 756) - transport_issues.diff uploaded by dvossel (license 671) Tested - by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: - https://reviewboard.asterisk.org/r/278/ ........ - -2009-06-16 16:34 +0000 [r200986] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged - revisions 200985 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | - kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 - lines Fix problems with new compiler attribute checking in - configure script. The last changes to ast_gcc_attribute.m4 caused - some problems checking for various attributes, because the scope - of the symbol the attribute is applied to can be important; this - patch allows the scope to be specified for the check. ........ - -2009-06-16 16:02 +0000 [r200945] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) - | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail - can only use one storage module at the moment. Because it's - unclear that selecting one of the storage modules in menuselect - will disable filesystem storage we now have a FILE_STORAGE option - that conflicts with the other modules. (closes issue #15333) - ........ - -2009-06-16 01:33 +0000 [r200724-200767] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, - autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 - Jun 2009) | 11 lines Ensure that configure-script testing for - compiler attributes actually works. The configure script tests - for compiler attributes didn't actually enable enough warnings or - provide a proper test harness to determine whether the compiler - supports the attribute in question or not; this caused gcc 4.1 to - report that it supports 'weakref', but it doesn't actually - support it in the way that is needed for our optional API - mechanism. The new configure script test will properly - distinguish between full support and partial support for this - attribute, among others. ........ - - * /, CHANGES: Merged revisions 200726 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | - kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 - lines Document the new automatic 'ignoresdpversion' behavior. - Asterisk will now automatically ignore incorrect incoming SDP - version numbers when necessary to complete a T.38 re-INVITE - operation. ........ - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 165180,200689 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | - mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 - lines This patch adds a new 'ignoresdpversion' option to - sip.conf. When this is enabled (either globally or for a specific - peer), chan_sip will treat any SDP data it receives as new data - and update the media stream accordingly. By default, Asterisk - will only modify the media stream if the SDP session version - received is different from the current SDP session version. This - option is required to interoperate with devices that have - non-standard SDP session version implementations (observed by toc - on the bug tracker with Microsoft OCS which always uses 0 as the - session version). http://reviewboard.digium.com/r/94/ (closes - issue #13958) Reported by: toc Tested by: toc ........ r200689 | - kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 - lines Accept T.38 re-INVITE responses with invalid SDP versions. - This commit changes the 'incoming SDP version' check logic a bit - more; when 'ignoresdpversion' is *not* set for a peer, if we - initiate a re-INVITE to switch to T.38, we'll always accept the - peer's SDP response, even if they don't properly increment the - SDP version number as they should. If this situation occurs, a - warning message will be generated suggesting that the peer's - configuration be changed to include the 'ignoresdpversion' - configuration option (although ideally they'd fix their SIP - implementation to be RFC compliant). AST-221 ........ - -2009-06-15 15:22 +0000 [r200515] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun - 2009) | 11 lines Merged revisions 200513 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun - 2009) | 5 lines Add INFO to our allowed methods so that endpoints - know they may send it to us. AST-223 ........ ................ - -2009-06-12 19:08 +0000 [r200362] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 200361 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun - 2009) | 16 lines Merged revisions 200360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun - 2009) | 10 lines Suppress a warning message and give a better - return code when generating inband ringing after a call is - answered. (closes issue #15158) Reported by: madkins Patches: - 15158.patch uploaded by mmichelson (license 60) Tested by: - madkins ........ ................ - -2009-06-11 22:42 +0000 [r200228] Sean Bright <sean@malleable.com> - - * Makefile, /: Merged revisions 199781 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | - seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 - lines Fix all of the parallel build warnings issued when running - make -j#. ........ - -2009-06-11 21:18 +0000 [r200149] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | - mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 - lines Fix a crash due to a potentially NULL p->options. Thanks to - mnicholson for pointing it out. ........ - -2009-06-11 12:16 +0000 [r200040] Leif Madsen <lmadsen@digium.com> - - * /, build_tools/make_version_c, build_tools/make_version_h: Merged - revisions 200039 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | - lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines - Fix path for .flavor and .version (issue #14737) Reported by: - davidw Patches: flavor.patch uploaded by davidw (license 780) - Tested by: davidw ........ - -2009-06-10 20:29 +0000 [r199994] David Brooks <dbrooks@digium.com> - - * main/pbx.c, /: Fixes the argument order in definition of - new_find_extension(). In the definition of new_find_extension(), - the arguments 'callerid' and 'label' were swapped. The prototype - declaration and all calls to the function are ordered 'callerid' - then 'label', but the function itself was ordered 'label' then - 'callerid'. (closes issue #15303) Reported by: JimDickenson - -2009-06-10 20:20 +0000 [r199975] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: The 1.6.0 branch was missing all - invite_branch logic. It has now been added. - -2009-06-10 16:13 +0000 [r199858] Sean Bright <sean@malleable.com> - - * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 - (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, - 10 Jun 2009) | 2 lines __WORDSIZE is not available on all - platforms, so use sizeof(void *) instead. ........ - ................ - -2009-06-09 20:54 +0000 [r199821] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | - dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines - CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command - only used UDP rather than copying the transport type from the - peer. (closes issue #15283) Reported by: jthurman Patches: - sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) - Tested by: jthurman, dvossel ........ - -2009-06-08 19:39 +0000 [r199632] Sean Bright <sean@malleable.com> - - * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 - (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun - 2009) | 21 lines Increase the size of our thread stack on 64 bit - processors. We were setting the stack size for each thread to - 240KB regardless of architecture, which meant that in some - scenarios we actually had less available stack space on 64 bit - processors (pointers use 8 bytes instead of 4). So now we - calculate the stack size we reserve based on the platform's - __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 - bit -> 1008KB (that's right, we're ready for 128 bit processors) - Patch typed by me but written by several members of - #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes - issue #14932) Reported by: jpiszcz Patches: - 06052009_issue14932.patch uploaded by seanbright (license 71) - Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 - 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the - stack size calculation just introduced. ........ ................ - -2009-06-05 21:37 +0000 [r199301] David Vossel <dvossel@digium.com> - - * main/pbx.c, /: Merged revisions 199298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) - | 21 lines Merged revisions 199297 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) - | 14 lines Fixes issue with hints giving unexpected results. - Hints with two or more devices that include ONHOLD gave - unexpected results. (closes issue #15057) Reported by: - p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel - (license 671) pbx.c.1.4.patch uploaded by p (license 558) - devicestate.c.trunk.patch uploaded by p (license 671) Tested by: - p_lindheimer, dvossel Review: - https://reviewboard.asterisk.org/r/254/ ........ ................ - -2009-06-05 13:51 +0000 [r199228] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun - 2009) | 14 lines Correct "dahdi show channels" output when - specifying a group. Since a DAHDI channel may belong to multiple - groups, we need to use a bitwise and instead of equivalence to - determine whether to display the channel information. (closes - issue #15248) Reported by: gentian Patches: 15248.patch uploaded - by mmichelson (license 60) Tested by: gentian ........ - -2009-06-04 19:16 +0000 [r199142] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 199139 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 - (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 - Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ - ................ - -2009-08-11 Tilghman Lesher <tlesher@digium.com> - - * Asterisk 1.6.0.13 released - - * channels/chan_sip.c: Bad merge from 1.6.0 branch - -2009-08-10 Tilghman Lesher <tlesher@digium.com> - - * Asterisk 1.6.0.12 released - - * AST-2009-005 - -2009-06-05 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.0.10 released - -2009-06-04 David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: Additional updates for AST-2009-001 - -2009-06-04 David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 - -2009-04-06 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.0.9 - -2009-04-03 16:27 -0500 [r186517] Mark Michelson <mmichelson@digium.com> - - * Remove an invalid call to free memory. - - A bad merge from trunk to 1.6.0 meant freeing memory that - should not be freed. In trunk, pkt->data is an ast_str, but - in 1.6.0, it is allocated in the same chunk of memory as the - sip_pkt. This only affects 1.6.0. - - (closes issue #14819) - Reported by: cwolff09 - -2009-04-02 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.0.8 - -2009-04-02 Tilghman Lesher <tlesher@digium.com> - - * Fix for security issue AST-2009-003 - -2009-03-30 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.0.7 - -2009-03-19 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.0.7-rc2 - -2009-03-19 15:40 +0000 [r183066-183109] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | - file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines - Improve our triggering of a T38 switchover internally when - triggered by a received reinvite. Previously we reached across - the channel bridge to get the other party's SIP dialog structure - in order to trigger an outgoing reinvite. This is extremely - dangerous to do and only works if bridged to another SIP channel. - This patch changes this to use the T38 control frame method of - requesting a switchover. This change also causes the SIP channel - driver to propogate back whether the switchover worked or not - instead of blindly accepting the incoming T38 reinvite. Review: - http://reviewboard.digium.com/r/200/ ........ - - * main/channel.c, /: Merged revisions 183057 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | - file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix - an issue where a T38 control frame would get dropped. If two - channels were bridged together using a generic bridge the T38 - control frame would get passed up instead of being indicated on - the other channel. ........ - -2009-03-18 21:19 +0000 [r183029] Jeff Peeler <jpeeler@digium.com> - - * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 - Mar 2009) | 4 lines Add some code removed by mistake from commit - 182722 that works around a file descriptor leak in versions of - PWLib prior to 1.12.0. ........ - -2009-03-18 14:24 +0000 [r182945] Russell Bryant <russell@digium.com> - - * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, - configure, apps/app_mp3.c, res/res_agi.c, - include/asterisk/poll-compat.h, channels/chan_alsa.c, - main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, - include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: - Merged revisions 182847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) - | 52 lines Merged revisions 182810 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) - | 44 lines Fix cases where the internal poll() was not being used - when it needed to be. We have seen a number of problems caused by - poll() not working properly on Mac OSX. If you search around, - you'll find a number of references to using select() instead of - poll() to work around these issues. In Asterisk, we've had poll.c - which implements poll() using select() internally. However, we - were still getting reports of problems. vadim investigated a bit - and realized that at least on his system, even though we were - compiling in poll.o, the system poll() was still being used. So, - the primary purpose of this patch is to ensure that we're using - the internal poll() when we want it to be used. The changes are: - 1) Remove logic for when internal poll should be used from the - Makefile. Instead, put it in the configure script. The logic in - the configure script is the same as it was in the Makefile. - Ideally, we would have a functionality test for the problem, but - that's not actually possible, since we would have to be able to - run an application on the _target_ system to test poll() - behavior. 2) Always include poll.o in the build, but it will be - empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() - throughout the source tree to ast_poll(). I feel that it is good - practice to give the API call a new name when we are changing its - behavior and not using the system version directly in all cases. - So, normally, ast_poll() is just redefined to poll(). On systems - where AST_POLL_COMPAT is defined, ast_poll() is redefined to - ast_internal_poll(). 4) Change poll() in main/poll.c to be - ast_internal_poll(). It's worth noting that any code that still - uses poll() directly will work fine (if they worked fine before). - So, for example, out of tree modules that are using poll() will - not stop working or anything. However, for modules to work - properly on Mac OSX, ast_poll() needs to be used. (closes issue - #13404) Reported by: agalbraith Tested by: russell, vadim - http://reviewboard.digium.com/r/198/ ........ ................ - -2009-03-17 20:51 +0000 [r182723] Jeff Peeler <jpeeler@digium.com> - - * channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, - configure, autoconf/ast_check_openh323.m4, - channels/h323/compat_h323.h, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged - revisions 182722 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | - jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines - Allow H.323 Plus library to be used in addition to the OpenH323 - library Chan_h323 can now be compiled against both the previously - supported versions of OpenH323 as well as the current H.323 Plus - (version 1.20.2). The configure script has been modified to look - in the default install location of h323 to hopefully help avoid - using the environment variables OPENH323DIR and PWLIBDIR. Also, - the CLI command "h323 show version" has been added which - indicates which version of h323 is in use. (closes issue #11261) - Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch - uploaded by jthurman (license 614) ........ - -2009-03-17 15:27 +0000 [r182569] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 182553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | - russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines - Tweak the handling of the frame list inside of ast_answer(). This - does not change any behavior, but moves the frames from the local - frame list back to the channel read queue using an O(n) algorithm - instead of O(n^2). ........ - -2009-03-17 15:00 +0000 [r182526-182532] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, /: Merged revisions 182530 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | - kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 - lines correct logic flaw in ast_answer() changes in r182525 - ........ - - * main/channel.c, /, main/features.c, include/asterisk/channel.h: - Merged revisions 182525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | - kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 - lines Improve behavior of ast_answer() to not lose incoming - frames ast_answer(), when supplied a delay before returning to - the caller, use ast_safe_sleep() to implement the delay. - Unfortunately during this time any incoming frames are discarded, - which is problematic for T.38 re-INVITES and other sorts of - channel operations. When a delay is not passed to ast_answer(), - it still delays for up to 500 milliseconds, waiting for media to - arrive. Again, though, it discards any control frames, or - non-voice media frames. This patch rectifies this situation, by - storing all incoming frames during the delay period on a list, - and then requeuing them onto the channel before returning to the - caller. http://reviewboard.digium.com/r/196/ ........ - -2009-03-17 05:53 +0000 [r182451] Tilghman Lesher <tlesher@digium.com> - - * main/db.c, /: Merged revisions 182450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) - | 14 lines Merged revisions 182449 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) - | 7 lines Fix race in astdb The underlying db1 implementation - does not fully isolate the pages retrieved from astdb, so the - lock protecting accesses needs to be extended until the copy from - the shared memory structure is done. (closes issue #14682) - Reported by: makoto ........ ................ - -2009-03-16 17:52 +0000 [r182283] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 182282 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 - (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) - | 7 lines Randomize IAX2 encryption padding The 16-32 byte random - padding at the beginning of an encrypted IAX2 frame turns out to - not be all that random at all. This patch calls ast_random to - fill the padding buffer with random data. The padding is - randomized at the beginning of every encrypted call and for every - encrypted retransmit frame. Review: - http://reviewboard.digium.com/r/193/ ........ ................ - -2009-03-16 17:36 +0000 [r182212-182279] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_env.c: Merged revisions 182278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 | - tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines - Fix an off-by-one error in the FILE() function, and extend - FILE()'s length parameter to work like variable substitution. - Previously, FILE() returned one less character than specified, - due to the terminating NULL. Both the offset and length - parameters now behave identically to the way variable - substitution offsets and lengths also work. (closes issue #14670) - Reported by: BMC ........ - - * channels/chan_local.c, /: Merged revisions 182211 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 - (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) - | 7 lines Fixup glare detection, to fix a memory leak of a local - pvt structure. (closes issue #14656) Reported by: caspy Patches: - 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) - Tested by: caspy ........ ................ - -2009-03-16 13:59 +0000 [r182172] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 182171 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | - file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix - a memory leak in the ast_answer / __ast_answer API call. For a - channel that is not yet answered this API call will wait until a - voice frame is received on the channel before returning. It does - this by waiting for frames on the channel and reading them in. - The frames read in were not freed when they should have been. - ........ - -2009-03-13 21:26 +0000 [r182064-182122] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 182121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | - mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 - lines Change faulty comparison used when announcing average hold - minutes and seconds (closes issue #14227) Reported by: caspy - ........ - - * /, main/features.c: Merged revisions 182029 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar - 2009) | 41 lines Merged revisions 181990 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar - 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and - peer when interpreting DTMF. Dynamic features defined in the - applicationmap section of features.conf allow one to specify - whether the caller, callee, or both have the ability to use the - feature. The documentation in the features.conf.sample file could - be interpreted to mean that one only needs to set the - DYNAMIC_FEATURES channel variable on the calling channel in order - to allow for the callee to be able to use the features which he - should have permission to use. However, the DYNAMIC_FEATURES - variable would only be read from the channel of the participant - that pressed the DTMF sequence to activate the feature. The - result of this was that the callee was unable to use dynamic - features unless the dialplan writer had taken measures to be sure - that the DYNAMIC_FEATURES variable was set on the callee's - channel. This commit changes the behavior of - ast_feature_interpret to concatenate the values of - DYNAMIC_FEATURES from both parties involved in the bridge. The - features themselves determine who has permission to use them, so - there is no reason to believe that one side of the bridge could - gain the ability to perform an action that they should not have - the ability to perform. Kevin Fleming pointed out on the - asterisk-users list that the typical way that this was worked - around in the past was by setting _DYNAMIC_FEATURES on the - calling channel so that the value would be inherited by the - called channel. While this works, the documentation alone is not - enough to figure out why this is necessary for the callee to be - able to use dynamic features. In this particular case, changing - the code to match the documentation is safe, easy, and will - generally make things easier for people for future installations. - This bug was originally reported on the asterisk-users list by - David Ruggles. (closes issue #14657) Reported by: mmichelson - Patches: 14657.patch uploaded by mmichelson (license 60) ........ - ................ - -2009-03-13 17:28 +0000 [r182036] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | - file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix - an issue with requesting a T38 reinvite before the call is - answered. The code responsible for sending the T38 reinvite did - not check if an INVITE was already being handled. This caused - things to get confused and the call to fail. The code now defers - sending the T38 reinvite until the current INVITE is done being - handled. (issue AST-191) ........ - -2009-03-12 21:44 +0000 [r181770-181848] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 181846 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | - mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 - lines Run the macro on the queue member's channel when he - answers, not the caller's channel. ........ - - * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar - 2009) | 28 lines Merged revisions 181768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar - 2009) | 22 lines Properly send a 487 on an INVITE we have not - responded to if we receive a BYE. If we receive an INVITE from an - endpoint and then later receive a BYE from that same endpoint - before we have sent a final response for the INVITE, then we need - to respond to the INVITE with a 487. There was logic in the code - prior to this commit which seemed to exist solely to handle this - situation, but there was one condition in an if statement which - was incorrect. The only way we would send a 487 was if the - sip_pvt had no owner channel. This made no sense since we created - the owner channel when we received the INVITE, meaning that the - majority of the time we would never send the 487. The 487 being - sent should not rely on whether we have created a channel. Its - delivery should be dependent on the current state of the initial - INVITE transaction. With this commit, that logic is now correctly - in place. (closes issue #14149) Reported by: legranjl Patches: - 14149.patch uploaded by mmichelson (license 60) Tested by: - legranjl ........ ................ - -2009-03-12 17:58 +0000 [r181732] Tilghman Lesher <tlesher@digium.com> - - * /, configure, main/translate.c: Merged revisions 181731 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 - Mar 2009) | 9 lines Adjust translation table column widths based - upon the translation times. Previously, only 5 columns were - displayed, and if a translation time exceeded 99,999 useconds, it - would be displayed as 0, instead of its actual time. (closes - issue #14532) Reported by: pj Patches: - 20090311__bug14532.diff.txt uploaded by tilghman (license 14) - Tested by: pj ........ - -2009-03-12 16:57 +0000 [r181613-181666] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, - 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 - lines Fix incorrect usage of strncasecmp... I really meant to use - strcasecmp. ........ ................ - - * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, - 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 - lines Fix another scenario where depending on configuration the - stream would not get read. For custom commands we don't know - whether the audio is coming from a stream or not so we are going - to have to read the data despite no channels. (closes issue - #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 - 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in - previous commit. ........ ................ - - * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, - 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | - 10 lines Fix issue with streaming MOH failing if nobody is - listening. When a music class is setup to actually provide music - on hold from a stream we need to constantly read audio from it - since it will constantly be providing audio. This is now done - despite there being no channels listening to it. (closes issue - #14416) Reported by: caspy ........ ................ - - * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | - file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix - crash when sleep and retries argument was not given to RetryDial - application. (closes issue #14647) Reported by: sherpya ........ - -2009-03-12 01:04 +0000 [r181543] Richard Mudgett <rmudgett@digium.com> - - * /, build_tools/make_version: Merged revisions 181542 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) - | 1 line Use the correct branch integrated property when - generating the version string ........ - -2009-03-11 23:19 +0000 [r181509] Michiel van Baak <michiel@vanbaak.info> - - * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk Provide - correct hint to debug SIP trouble in the default config (closes - issue #14646) Reported by: strk - -2009-03-11 22:22 +0000 [r181450] Jason Parker <jparker@digium.com> - - * /, configure, configure.ac: Merged revisions 181444 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 - (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | - 4 lines Allow prefix to set localstatedir (when used and - different from the default). This is similar to the /etc change - that was made for the non-FreeBSD case. ........ ................ - -2009-03-11 22:15 +0000 [r181425-181429] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 181428 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | - russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines - Make handling of the BRIDGEPVTCALLID variable thread-safe. - ........ - - * main/channel.c, /: Merged revisions 181424 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) - | 17 lines Merged revisions 181423 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) - | 9 lines Make code that updates BRIDGEPEER variable thread-safe. - It is not safe to read the name field of an ast_channel without - the channel locked. This patch fixes some places in channel.c - where this was being done, and lead to crashes related to - masquerades. (closes issue #14623) Reported by: guillecabeza - ........ ................ - -2009-03-11 17:37 +0000 [r181372] David Vossel <dvossel@digium.com> - - * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions - 181371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) - | 17 lines Merged revisions 181340 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) - | 11 lines encrypted IAX2 during packet loss causes decryption to - fail on retransmitted frames If an iax channel is encrypted, and - a retransmit frame is sent, that packet's iseqno is updated while - it is encrypted. This causes the entire frame to be corrupted. - When the corrupted frame is sent, the other side decrypts it and - sends a VNAK back because the decrypted frame doesn't make any - sense. When we get the VNAK, we look through the sent queue and - send the same corrupted frame causing a loop. To fix this, - encrypted frames requiring retransmission are decrypted, updated, - then re-encrypted. Since key-rotation may change the key held by - the pvt struct, the keys used for encryption/decryption are held - within the iax_frame to guarantee they remain correct. (closes - issue #14607) Reported by: stevenla Tested by: dvossel Review: - http://reviewboard.digium.com/r/192/ ........ ................ - -2009-03-11 17:28 +0000 [r181297-181352] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | - 21 lines Merged revisions 181328 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | - 14 lines Fix issue where an attended transfer could not be - completed under a rare scenario. When completing an attended - transfer chan_sip does a check to make sure the extension in the - URI portion of the Refer-To header is a local valid extension. We - don't actually need to check this since we know for sure the - other channel is already up and talking to the extension. Some - devices do not put the extension in the Refer-To header either, - which can cause the extension check to fail. We now no longer do - this check if it is an attended transfer. (closes issue #14628) - Reported by: sverre Patches: 14628.diff uploaded by file (license - 11) ........ ................ - - * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | - 16 lines Merged revisions 181295 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 - lines Fix a problem with inband DTMF detection on outgoing SIP - calls when dtmfmode=auto. When dtmfmode was set to auto the - inband DTMF detector was not setup on outgoing SIP calls. This - caused inband DTMF detection to fail. The inband DTMF detector is - now setup for both dtmfmode inband and auto. (closes issue - #13713) Reported by: makoto ........ ................ - -2009-03-11 15:54 +0000 [r181137-181284] Jeff Peeler <jpeeler@digium.com> - - * channels/h323/ast_h323.cxx: add missing header file - - * utils/extconf.c: Fix merge oops from 181137 - - * utils/Makefile, include/asterisk/utils.h, - include/asterisk/astmm.h, /, channels/chan_sip.c, - channels/h323/ast_h323.cxx, utils/extconf.c: Merged revisions - 181135 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | - jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines - Fix malloc debug macros to work properly with h323. The main - problem here was that cstdlib was undefining free thereby causing - the proper debug macros to not be used. ast_h323.cxx has been - changed to call ast_free instead to avoid the issue. A few other - issues were addressed: - There were a few instances of functions - improperly passing ast_free instead of ast_free_ptr. - Some clean - up was done to avoid the debug macros intentionally being - redefined. (copied below from Kevin's commit, appreciate the - help) - disable astmm.h from doing anything when STANDALONE is - defined, which is used by the tools in the utils/ directory that - use parts of Asterisk header files in hackish ways; also ensure - that utils/extconf.c and utils/conf2ael.c are compiled with - STANDALONE defined. (closes issue #13593) Reported by: pj - ........ - -2009-03-11 00:52 +0000 [r181034] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 181032-181033 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 - (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar - 2009) | 9 lines Fix incorrect tag checking on transfers when - pedantic=yes is enabled. (closes issue #14611) Reported by: - klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt - uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ - r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar - 2009) | 3 lines Remove unused variables. ........ - ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 - (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC - 3891 ................ - -2009-03-10 22:05 +0000 [r180946] Jason Parker <jparker@digium.com> - - * /, configure, configure.ac, autoconf/ast_prog_sed.m4, - autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 - (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar - 2009) | 1 line Make things happier when using autoconf 2.62+ - ........ ................ - -2009-03-10 14:41 +0000 [r180718-180801] Joshua Colp <jcolp@digium.com> - - * main/manager.c, /: Merged revisions 180800 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | - file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines - Reset the thread local string buffer when handling the UserEvent - action. (closes issue #14593) Reported by: JimDickenson ........ - - * channels/chan_sip.c: If a port is specified when dialing a peer - then use it. (closes issue #14626) Reported by: acunningham - - * channels/chan_sip.c: Ensure that the new outgoing dialog to a - peer is able to set the socket details, even if the default is - present. (closes issue #14480) Reported by: jon - -2009-03-06 18:26 +0000 [r180582] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 - (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, - 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when - IMAP storage is enabled. ........ ................ - -2009-03-06 Leif Madsen <lmadsen@digium.com> - - * Release 1.6.0.7-rc1 - -2009-03-06 17:28 +0000 [r180535] David Vossel <dvossel@digium.com> - - * main/enum.c, /: Merged revisions 180534 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) - | 15 lines Merged revisions 180532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) - | 9 lines Fix handling of backreferences for ENUM lookups enum.c - did not handle regex backtraces correctly. The '\1' in the regex - is a backreference that requires a pattern match to be inserted. - The way the code used to work is that it would find the - backreference and insert the entire input string minus the '+'. - This is incorrect. The regexec() function takes in a variable - called pmatch which is an array of structs containing the start - and end indexes for each backreference substring. The original - code actually passed the pmatch array pointer into regexec but - never did anything with it. Now when a backtrace is found, the - backtrace number is looked up in the pmatch array and the correct - substring is inserted. (closes issue #14576) Reported by: - chris-mac Review: http://reviewboard.digium.com/r/187/ ........ - ................ - -2009-03-05 23:28 +0000 [r180404-180466] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 - (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar - 2009) | 16 lines [IMAP] Fix message retrieval issues when - identical mailbox names were defined in separate contexts. There - was a fix put in a while back so that an X-Asterisk-VM-Context - message header was added to stored IMAP voicemails. This would - allow for us to differentiate if the same mailbox name was used - in multiple contexts. The problem still left was that not all - places where messages were retrieved actually attempted to use - this header for information when retrieving messages. This commit - fixes that so that MWI and message retrieval from VoiceMailMain - work as expected. (closes issue #13853) Reported by: vicks1 - Patches: 13853_v2.patch uploaded by mmichelson (license 60) - Tested by: lmadsen ........ ................ - - * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged - revisions 180383 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar - 2009) | 31 lines Merged revisions 180380 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar - 2009) | 25 lines Fix broken mailbox parsing when searchcontexts - option is enabled. When using the searchcontexts option in - voicemail.conf, the code made the assumption that all mailbox - names defined were unique across all contexts. However, the code - did nothing to actually enforce this assumption, nor did it do - anything to alert a user that he may have created an ambiguity in - his voicemail.conf file by defining the same mailbox name in - multiple contexts. With this change, we now will issue a nice - long warning if searchcontexts is on and we encounter the same - mailbox name in multiple contexts and ignore any duplicates after - the first box. Whether searchcontexts is enabled or not, if we - come across a duplicate mailbox in the same context, then we will - issue a warning and ignore the duplicated mailbox. I have also - added a small note to voicemail.conf.sample in the explanation - for searchcontexts explaining that you cannot define the same - mailbox in multiple contexts if you have enabled the option. - (closes issue #14599) Reported by: lmadsen Patches: 14599.patch - uploaded by mmichelson (license 60) (with slight modification) - Tested by: lmadsen ........ ................ - -2009-03-05 18:36 +0000 [r180377] Kevin P. Fleming <kpfleming@digium.com> - - * main/rtp.c, main/frame.c, /, include/asterisk/frame.h: Merged - revisions 180373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar - 2009) | 15 lines Merged revisions 180372 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar - 2009) | 9 lines Fix problems when RTP packet frame size is - changed During some code analysis, I found that calling - ast_rtp_codec_setpref() on an ast_rtp session does not work as - expected; it does not adjust the smoother that may on the RTP - session, in fact it summarily drops it, even if it has data in - it, even if the current format's framing size has not changed. - This is not good. This patch changes this behavior, so that if - the packetization size for the current format changes, any - existing smoother is safely updated to use the new size, and if - no smoother was present, one is created. A new API call for - smoothers, ast_smoother_reconfigure(), was required to implement - these changes. Review: http://reviewboard.digium.com/r/184/ - ........ ................ - -2009-03-04 19:25 +0000 [r180121-180196] Joshua Colp <jcolp@digium.com> - - * /, main/callerid.c: Merged revisions 180195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | - 11 lines Merged revisions 180194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 - lines Look for the number in a callerid string starting from the - end. This way a value using <> can exist in the name portion. - (issue #AST-194) ........ ................ - - * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | - file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines - Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) - Reported by: alecdavis Patches: app_dial.optionk.diff.txt - uploaded by alecdavis (license 585) ........ - -2009-03-03 23:35 +0000 [r180078] David Vossel <dvossel@digium.com> - - * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, - main/app.c: Merged revisions 180032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | - dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines - app_read does not break from prompt loop with user terminated - empty string In app.c, ast_app_getdata is called to stream the - prompts and receive DTMF input. If ast_app_getdata() receives an - empty string caused by the user inputing the end of string - character, in this case '#', it should break from the prompt loop - and return to app_read, but instead it cycles through all the - prompts. I've added a return value for this special case in - ast_readstring() which uses an enum I've delcared in apps.h. This - enum is now used as a return value for ast_app_getdata(). (closes - issue #14279) Reported by: Marquis Patches: fix_app_read.patch - uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded - by dvossel (license 671) Tested by: Marquis, dvossel Review: - http://reviewboard.digium.com/r/177/ ........ - -2009-03-03 23:26 +0000 [r180058] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, - utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, - main/ast_expr2f.c: Merged revisions 179973 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | - 33 lines Merged revisions 179807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some - work to do to port these changes to trunk; the check_expr stuff - hasn't been updated here for quite some time, it appears. I added - some more tests to the check_expr2 suite. I had to play around - with the makefile a bit, etc. I added STANDALONE2 #ifdefs to - ast_expr2.y so as not to conflict structure with aelparse. - ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar - 2009) | 19 lines These changes allow AEL to better check ${} - constructs within $[...], that are concatenated with text. I - modified and added rules in ast_expr2.fl to better handle the - concatenations. I added some default routines to ast_expr2.y so - the standalone would compile. It also looks like I haven't run - this thru bison since 2.1, so it's good to get this updated. The - Makefile has comments added now for check_expr2 and check_expr to - explain what they are for, and how to run them. The testexpr2s - stuff has been removed, in favor of check_expr2. expr2.testinput - has been updated to include the two expressions that inspired - these changes (from mcnobody on #asterisk this morning) The - regression has been run and all looks well. ........ - ................ - -2009-03-03 22:49 +0000 [r179971-180008] Mark Michelson <mmichelson@digium.com> - - * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions - 180007 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar - 2009) | 22 lines Merged revisions 180006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar - 2009) | 17 lines Clarify some documentation of queues.conf.sample - It had always been possible to explicitly specify a "blank" value - for a sound file in queues.conf and have no sound played back. - The problem with this is that it would result in some ugly CLI - warnings from file.c. This commit introduces a check when playing - a file in app_queue to see if the name of the file is zero-length - and return early if that is the case. Also, the ability to - specify the blank sound files in queues.conf is now mentioned - more clearly in queues.conf.sample (closes issue #14227) Reported - by: caspy ........ ................ - - * apps/app_queue.c: Fix a memory leak when updating a realtime - member field. This was discovered while looking at issue #14353 - -2009-03-03 18:29 +0000 [r179842] Joshua Colp <jcolp@digium.com> - - * /, main/features.c: Merged revisions 179841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | - 16 lines Merged revisions 179840 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 - lines Do not assume that the bridge_cdr is still attached to the - channel when the 'h' exten is finished executing. It is possible - for a masquerade operation to occur when the 'h' exten is - operating. This operation moves the CDR records around causing - the bridge_cdr to no longer exist on the channel where it is - expected to. We can not safely modify it afterwards because of - this, so don't even try. (closes issue #14564) Reported by: meric - ........ ................ - -2009-03-03 16:48 +0000 [r179743] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 179742 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) - | 14 lines Merged revisions 179741 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) - | 6 lines Ensure chan->fdno always gets reset to -1 after - handling a channel fd event. Since setting fdno to -1 had to be - moved, a couple of other code paths that do process an fd event - return early and do not pass through the code path where it was - moved to. So, set it to -1 in a few other places, too. ........ - ................ - -2009-03-03 14:40 +0000 [r179673] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 179672 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | - 10 lines Merged revisions 179671 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 - lines Move where fdno is set to the default value to *after* the - read callback of the channel driver is called. We have to do this - as the underlying channel driver may need the fdno value to - determine what to read. ........ ................ - -2009-03-03 13:55 +0000 [r179610] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 179609 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) - | 17 lines Merged revisions 179608 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) - | 9 lines Make it easier to detect an improper call to - ast_read(). When you call ast_waitfor() on a channel, the index - into the channel fds array that holds the file descriptor that - poll() determines has input available is stored in fdno. This - patch clears out this value after a call to ast_read() and also - reports errors if ast_read() is called without an fdno set. From - a discussion on the asterisk-dev list. ........ ................ - -2009-03-03 00:03 +0000 [r179538] Jeff Peeler <jpeeler@digium.com> - - * main/channel.c, /: Merged revisions 179537 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) - | 21 lines Merged revisions 179536 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) - | 15 lines Fix bridging regression from commit 176701 This fixes - a bad regression where the bridge would exit after an attended - transfer was made. The problem was due to nexteventts getting set - after the masquerade which caused the bridge to return - AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: - tim_ringenbach ........ ................ - -2009-03-02 23:38 +0000 [r179534] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) - | 48 lines Merged revisions 179532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) - | 40 lines Move ast_waitfor() down to avoid the results of the - API call becoming stale. This call to ast_waitfor() was being - done way too soon in this section of code. Specifically, there - was code in between the call to waitfor and the code that uses - the result that puts the channel in autoservice. By putting the - channel in autoservice, the previous results of ast_waitfor() - become meaningless, as the autoservice thread will do it's own - ast_waitfor() and ast_read() on the channel. So, when we came - back out of autoservice and eventually hit the block of code that - calls ast_read() on the channel, there may not actually be any - input on the channel available. Even though the previous call to - ast_waitfor() in app_meetme said there was input, the autoservice - thread has since serviced the channel for some period of time. - This bug manifested itself while dvossel was doing some testing - of MeetMe in Asterisk trunk. He was using the timerfd timing - module. When the code hit ast_read() erroneously, it determined - that it must have been called because of input on the timer fd, - as chan->fdno was set to AST_TIMING_FD, since that was the cause - of the last legitimate call to ast_read() done by autoservice. In - this test, an IAX2 channel was calling into the MeetMe - conference. It was _much_ more likely to be seen with an IAX2 - channel because of the way audio is handled. Every audio frame - that comes in results in a call to ast_queue_frame(), which then - uses ast_timer_enable_continuous() to notify the channel thread - that a frame is waiting to be handled. So, the chances of - ast_waitfor() indicating that a channel needs servicing due to a - timer event on an IAX2 event is very high. Finally, it is - interesting to note that if a different timing interface was - being used, this bug would probably not be noticed. When - ast_read() is called and erroneously thinks that there is a timer - event to handle, it calls the ast_timer_ack() function. The - pthread and dahdi timing modules handle the ack() function being - called when there is no event by simply ignoring it. In the case - of the timerfd module, it results in a read() on the timer fd - that will block forever, as there is no data to read. This caused - Asterisk to lock up very quickly. Thanks to dvossel and - mmichelson for the fun debugging session. :-) ........ - ................ - -2009-03-02 23:15 +0000 [r179473] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 151464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | - mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 - lines Make the sip_standard_port function more granular by - allowing separate type and port arguments. This is necessary - because when building our From and Contact headers, we need to be - absolutely sure that we are placing our source port there and not - the peer's source port. (closes issue #12761) Reported by: - asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by - asbestoshead (license 455) ........ - -2009-03-02 23:11 +0000 [r179470] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: Merged revisions 179469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) - | 17 lines Merged revisions 179468 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) - | 10 lines When ending a recording with silence detection, - remember to reduce the duration. The end of the recording is - correspondingly trimmed, but the duration was not trimmed by the - number of seconds trimmed, so the saved duration was necessarily - longer than the actual soundfile duration. (closes issue #14406) - Reported by: sasargen Patches: 20090226__bug14406.diff.txt - uploaded by tilghman (license 14) Tested by: sasargen ........ - ................ - -2009-03-02 23:02 +0000 [r179463] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 179462 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) - | 16 lines Merged revisions 179461 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) - | 8 lines Ensure that only one thread is calling ast_settimeout() - on a channel at a time. For example, with an IAX2 channel, you - can have both the channel thread and the chan_iax2 processing - threads calling this function, and doing so twice at the same - time is a bad thing. (Found in a debugging session with dvossel - and mmichelson) ........ ................ - -2009-03-02 20:17 +0000 [r179402] Jason Parker <jparker@digium.com> - - * /, main/editline/configure, main/editline/np/unvis.c, - main/editline/sys.h, main/editline/configure.in: Merged revisions - 179396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | - 9 lines Merged revisions 179395 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | - 1 line Remove several silly warnings in editline. One about a - broken preprocessor directive, and another about strlcpy/strlcat. - (closes issue #14264) Reported by: dimas ........ - ................ - -2009-03-02 17:58 +0000 [r179360-179363] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: KeepAlive application no longer exists, so fix - gosub implementation to not use it. (closes issue #14571) - Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded - by tilghman (license 14) Tested by: tilghman - - * cdr/cdr_sqlite3_custom.c: If cdr registration somehow succeeds - without a config file, don't crash. (closes issue #14563) - Reported by: alerios - -2009-03-01 22:07 +0000 [r179220-179222] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add error checking when updating the "paused" - field of a realtime queue member. This code already existed in - trunk and 1.6.1, but was not in 1.6.0 prior to this commit. - (closes issue #14338) Reported by: fiddur Patches: 14338.patch - uploaded by mmichelson (license 60) Tested by: fiddur - - * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | - mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 - lines Properly free memory and remove scheduler entries when a - transmission failure occurs. Previously, only the "data" field of - the sip_pkt created during __sip_reliable_xmit was freed when - XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was - called, this inevitably resulted in the reading and writing of - freed memory. XMIT_FAILURE is a condition meaning that we don't - want to attempt resending the packet at all. The proper action to - take is to remove the scheduler entry we just created, free the - packet's data as well as the packet itself, and unlink it from - the list of packets on the sip_pvt structure. (closes issue - #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by - mmichelson (license 60) Tested by: Nick_Lewis ........ - -2009-02-27 21:33 +0000 [r179162] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) - | 3 lines If config file is blank, don't load module. (Closes - issue #14563) ........ - -2009-02-27 19:05 +0000 [r179058] Jason Parker <jparker@digium.com> - - * /, doc/tex/channelvariables.tex: Merged revisions 179057 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb - 2009) | 8 lines Update documentation for DIALEDTIME and - ANSWEREDTIME variables. (closes issue #14566) Reported by: - klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by - klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by - klaus3000 (license 65) ........ - -2009-02-27 03:52 +0000 [r178987] Steve Murphy <murf@digium.com> - - * configs/features.conf.sample, /, main/features.c: Merged - revisions 178986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | - 26 lines Merged revisions 178956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 In this - case, it's just a matter of reducing the default timeouts from - 2000 to 1000 msec, as the max def feature digit timeout is no - longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 - -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default - feature digit timeout to 1000 ms from the previous default of - 500. As per bug 14515, a dev discussion arrived at a "mediated - concensus" of a default feature digit timeout of 1.0 sec. Some - voted for 1300; ctooley thought 1500 for distracted phone users - in phone booths; kpfleming put his foot down at 1.0 sec. Users - who found the previous default max delay of 250 msec perfect, are - welcome to override the new default. Notice that I said that 250 - msec was the default; wait a minute, you might say, the config - file said it was 500 msec!; well, because of the bug fix for - 14515, we found that 500 msec was actually enforcing a max of - 250. The bug fix would restore 500 msec, but we felt even that - was a bit tight for most users... 2000 msec was pushed earlier by - mmichelson, so that reduces to 1000 msec after the bug fix. - Enjoy! ........ ................ - -2009-02-26 17:50 +0000 [r178874] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 178871 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) - | 6 lines IAX2 prune realtime, minor tweak to last fix A return - statement was missing which caused unexpected cli output. issue - #14479 ........ - -2009-02-26 17:29 +0000 [r178866] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 178828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | - 34 lines Merged revisions 178804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | - 28 lines This patch prevents the feature detection timeout from - being cut in half. Because the ast_channel_bridge() call will - return 0 and pass a frame pointer for both DTMF_BEGIN and - DTMF_END, the feature_timer field in hte config struct is getting - decremented twice, which effectively cuts the digittimeout in - half. I added conditions to the if statement to only let DTMF_END - frames to flow thru, which solved the problem. Also, when the - frame pointer is null, let control flow thru-- this usually - happens on timeouts. I added a comment to the code to explain - what's going on and why. Many thanks to sodom for reporting this - problem. Personnally, it always seemed like something was wrong - with the featuredigittimeout, but I never could quite decide - what... and was too busy to investigate. This bug forced the - issue, and now we know. Sodom had other issues in 14515, but I - couldn't reproduce them. If he still has problems, and wants to - get them solved, he is welcome to reopen 14515. (closes issue - #14515) Reported by: sodom Patches: 14515.patch uploaded by murf - (license 17) Tested by: murf, sodom ........ ................ - -2009-02-26 16:01 +0000 [r178768] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 178767 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) - | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. - If "iax2 prune realtime all" was called, it would appear like the - command was successful, but in reality nothing happened. This is - because the reload that was supposed to take place checks the - config files, sees no changes, and does nothing. If there had - been a change in the the config file, the realtime users would - have been marked for deletion and everything would have been - fine. Now prune_users() and prune_peers() are called instead of - reload_config() to prune all users/peers that are realtime. These - functions remove all users/peers with the rtfriend and delme - flags set. iax2_prune_realtime() also lacked the code to properly - delete a single friend. For example. if iax2 prune realtime - <friend> was called, only the peer instance would be removed. The - user would still remain. (closes issue #14479) Reported by: - mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ - -2009-02-25 12:46 +0000 [r178510] Russell Bryant <russell@digium.com> - - * main/asterisk.c, /: Merged revisions 178509 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) - | 10 lines Merged revisions 178508 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) - | 2 lines Update the copyright year for the main page of the - doxygen documentation. ........ ................ - -2009-02-24 23:28 +0000 [r178382-178447] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 178446 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 - (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) - | 5 lines Add section about the #exec command in configuration - files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, - with additional notes by tilghman (license 14) ........ - ................ - - * main/asterisk.c, /: Merged revisions 178381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | - tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines - Apparently, a void cast doesn't override warn_unused_result. - ........ - -2009-02-24 20:43 +0000 [r178378] Russell Bryant <russell@digium.com> - - * main/rtp.c, /: Merged revisions 178374 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) - | 14 lines Merged revisions 178373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) - | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset - to 0 properly. (issue #14460) Reported by: moliveras Tested by: - russell ........ ................ - -2009-02-24 20:40 +0000 [r178343-178376] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 178375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | - tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines - The 3 possible errors with pipe(2) are all impossible in this - situation. ........ - - * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 - Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of - depending upon the astcanary process being inherited by init. - ........ - -2009-02-24 18:05 +0000 [r178306] Terry Wilson <twilson@digium.com> - - * apps/app_dahdiras.c: Change include order to make compile on - Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before - linux/types.h is included, the __s32 type doesn't get defined - -2009-02-24 17:53 +0000 [r178304] Tilghman Lesher <tlesher@digium.com> - - * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | - tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines - Cause astcanary to exit if Asterisk exits abnormally and doesn't - kill astcanary. Also, add some documentation supporting the use - of astcanary. (closes issue #14538) Reported by: KNK Patches: - asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) - ........ - -2009-02-24 15:20 +0000 [r178224] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | - 16 lines Merged revisions 178205 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 - lines Skip check for extension when subscribing for MWI. Since - the remote side is not actually subscribing to a specific - extension when subscribing for MWI just skip the check to see if - the extension exists. They can't use it to specify the mailbox - either since we require configuration of that in sip.conf (closes - issue #14531) Reported by: festr ........ ................ - -2009-02-23 23:17 +0000 [r178145] Russell Bryant <russell@digium.com> - - * main/rtp.c, /: Merged revisions 178142 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) - | 22 lines Merged revisions 178141 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) - | 14 lines Fix infinite DTMF when a BEGIN is received without an - END. This commit is related to rev 175124 of 1.4 where a previous - attempt was made to fix this problem. The problem with the - previous patch was that the inserted code needed to go _before_ - setting the lastrxts to the current timestamp. Because those were - the same, the dtmfcount variable was never decremented, and so - the END was never sent. In passing, I removed the dtmfsamples - variable which was completed unused. I also removed a redundant - setting of the lastrxts variable. (closes issue #14460) Reported - by: moliveras ........ ................ - -2009-02-23 Leif Madsen <lmadsen@digium.com> - - * Released 1.6.0.6 - -2009-02-13 Leif Madsen <lmadsen@digium.com> - - * Released 1.6.0.6-rc1 - -2009-02-13 16:43 +0000 [r175550] Joshua Colp <jcolp@digium.com> - - * /, apps/app_record.c: Merged revisions 175549 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | - file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add - an option to keep the recorded file upon hangup. (closes issue - #14341) Reported by: fnordian ........ - -2009-02-12 21:41 +0000 [r175369] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | - russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines - Remove useless string copy, and make sscanf safe again ........ - -2009-02-12 21:27 +0000 [r175347] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, /: Merged revisions 175334 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) - | 16 lines Merged revisions 175311 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) - | 9 lines Fix crashes when receiving certain T.38 packets. Also, - increase the maximum size of T.38 packets and warn users when - they try to set the limits above those maximums. (closes issue - #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt - uploaded by Corydon76 (license 14) Tested by: schern ........ - ................ - -2009-02-12 20:59 +0000 [r175299-175301] Jeff Peeler <jpeeler@digium.com> - - * main/features.c: Fix mistake in merging conflict from 175299. - - * /, main/features.c: Merged revisions 175298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) - | 15 lines Merged revisions 175294 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) - | 9 lines Fix ParkedCall event information for From field in the - case of a blind transfer If the parker information can not be - obtained from the peer, try and see if the BLINDTRANSFER channel - variable has been set. Previously, a blind transfer to the - ParkAndAnnounce app would return nothing for the From. Closes - AST-189 ........ ................ - -2009-02-12 20:46 +0000 [r175256-175296] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | - russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines - Avoid using ast_strdupa() in a loop. ........ - - * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) - | 4 lines Don't enable something by default that has a dependency - on something _not_ enabled by default. menuselect was not happy - with this. ........ - -2009-02-12 18:00 +0000 [r175189] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c: Merged revisions 175188 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) - | 12 lines Merged revisions 175187 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) - | 6 lines Fix crash in event of failed attempt to transfer to - parking The peer may not necessarily exist, such as in the case - of a transfer to ParkAndAnnounce. In this case don't try to play - a sound to it. ........ ................ - -2009-02-12 17:03 +0000 [r175126] Russell Bryant <russell@digium.com> - - * main/rtp.c, /: Merged revisions 175125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) - | 35 lines Merged revisions 175124 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) - | 27 lines Don't send DTMF for infinite time if we do not receive - an END event. I thought that this was going to end up being a - pretty gnarly fix, but it turns out that there was actually - already a configuration option in rtp.conf, dtmftimeout, that was - intended to handle this situation. However, in between Asterisk - 1.2 and Asterisk 1.4, the code that processed the option got - lost. So, this commit brings it back to life. The default timeout - is 3 seconds. However, it is worth noting that having this be - configurable at all is not really the recommended behavior in RFC - 2833. From Section 3.5 of RFC 2833: Limiting the time period of - extending the tone is necessary to avoid that a tone "gets - stuck". Regardless of the algorithm used, the tone SHOULD NOT be - extended by more than three packet interarrival times. A slight - extension of tone durations and shortening of pauses is generally - harmless. Three seconds will pretty much _always_ be far more - than three packet interarrival times. However, that behavior is - not required, so I'm going to leave it with our legacy behavior - for now. Code from svn/asterisk/team/russell/issue_14460 (closes - issue #14460) Reported by: moliveras ........ ................ - -2009-02-12 16:33 +0000 [r175122] Mark Michelson <mmichelson@digium.com> - - * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions - 175121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | - mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 - lines Make lock information for ao2_trylock be more useful and - gnarly Core show locks information involving an ao2_trylock did - not show the function that called ao2_trylock, but would instead - show ao2_trylock as the source of the lock. This is not useful - when trying to debug locking issues. One bizarre note is that - this logic is already in 1.4 but somehow did not get merged to - trunk or the 1.6.X branches. ........ - -2009-02-12 14:27 +0000 [r175059-175090] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) - | 6 lines Issue a warning message if our candidate's IP is the - loopback address. (closes issue #13985) Reported by: jcovert - Tested by: phsultan ........ - - * /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 - (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) - | 12 lines Set the initiator attribute to lowercase in our - replies when receiving calls. This attribute contains a JID that - identifies the initiator of the GoogleTalk voice session. The - GoogleTalk client discards Asterisk's replies if the initiator - attribute contains uppercase characters. (closes issue #13984) - Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by - jcovert (license 551) Tested by: jcovert ........ - ................ - -2009-02-11 23:04 +0000 [r174765-174949] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 174948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb - 2009) | 35 lines Fix odd "thank you" sound playing behavior in - app_queue.c If someone has configured the queue to play an - position or holdtime announcement, then it is odd and potentially - unexpected to hear a "Thank you for your patience" sound when no - position or holdtime was actually announced. This fixes the - announcement so that the "thanks" sound is only played in the - case that a position or holdtime was actually announced. There is - a way that the "thank you" sound can be played without a position - or holdtime, and that is to set announce-frequency to a value but - keep announce-position and announce-holdtime both turned off. - (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch - uploaded by putnopvut (license 60) Tested by: caspy - ................ - - * apps/app_dial.c, main/channel.c, main/pbx.c, /, - apps/app_dictate.c, apps/app_waitforsilence.c, - include/asterisk/channel.h: Merged revisions 174945 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb - 2009) | 29 lines Fix 'd' option for app_dial and add new option - to Answer application The 'd' option would not work for channel - types which use RTP to transport DTMF digits. The only way to - allow for this to work was to answer the channel if we saw that - this option was enabled. I realized that this may cause issues - with CDRs, specifically with giving false dispositions and answer - times. I therefore modified ast_answer to take another parameter - which would tell if the CDR should be marked answered. I also - extended this to the Answer application so that the channel may - be answered but not CDRified if desired. I also modified - app_dictate and app_waitforsilence to only answer the channel if - it is not already up, to help not allow for faulty CDR answer - times. All of these changes are going into Asterisk trunk. For - 1.6.0 and 1.6.1, however, all the changes except for the change - to the Answer application will go in since we do not introduce - new features into stable branches (closes issue #14164) Reported - by: DennisD Patches: 14164.patch uploaded by putnopvut (license - 60) Tested by: putnopvut Review: - http://reviewboard.digium.com/r/145 ........ - - * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | - mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 - lines Fix potential for stack overflows in app_chanspy.c When - using the 'g' or 'e' options, the stack allocations that were - used could cause a stack overflow if a spyer stayed on the line - long enough without actually successfully spying on anyone. The - problem has been corrected by using static buffers and copying - the contents of the appropriate strings into them instead of - using functions like alloca or ast_strdupa ........ - - * main/manager.c, /: Merged revisions 174764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | - mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 - lines Fix an fd leak that would occur in HTTP AMI sessions The - explanation behind this fix is a bit complicated, and I've - already typed it up in the code as a huge comment inside of - manager.c, so I'll give the abridged version here. We needed a - way to separate action-specific data from session-specific data. - Unfortunately, the only way to maintain API compatibility and to - not have to change every single manager action was to rename the - current mansession structure and wrap it inside a new mansession - structure which actually contains action- specific data. (closes - issue #14364) Reported by: awk Patches: 14364_better.patch - uploaded by putnopvut (license 60) Tested by: putnopvut Review: - http://reviewboard.digium.com/r/148/ ........ - -2009-02-10 20:16 +0000 [r174711] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | - file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines - Only decrease inringing count if above zero. (issue #13238) - Reported by: kowalma ........ - -2009-02-10 18:19 +0000 [r174596] Matthew Nicholson <mnicholson@digium.com> - - * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb - 2009) | 25 lines Merged revisions 174583 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb - 2009) | 18 lines Improve behavior of jitterbuffer when - maxjitterbuffer is set. This change improves the way the - jitterbuffer handles maxjitterbuffer and dramatically reduces the - number of frames dropped when maxjitterbuffer is exceeded. In the - previous jitterbuffer, when maxjitterbuffer was exceeded, all new - frames were dropped until the jitterbuffer is empty. This change - modifies the code to only drop frames until maxjitterbuffer is no - longer exceeded. Also, previously when maxjitterbuffer was - exceeded, dropped frames were not tracked causing stats for - dropped frames to be incorrect, this change also addresses that - problem. (closes issue #14044) Patches: bug14044-1.diff uploaded - by mnicholson (license 96) Tested by: mnicholson Review: - http://reviewboard.digium.com/r/144/ ........ ................ - -2009-02-10 15:39 +0000 [r174544] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | - file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines - Make the logic for inuse and inringing manipluation match that of - 1.4. The old broken logic would reset the values back to 0 during - certain scenarios causing the wrong state to be reported. (closes - issue #14399) Reported by: caspy (issue #13238) Reported by: - kowalma ........ - -2009-02-10 05:06 +0000 [r174439] Steve Murphy <murf@digium.com> - - * apps/app_rpt.c: For some strange reason, I didn't think 1.6.0 - needed this fix. I was wrong. Here it is. - -2009-02-09 17:28 +0000 [r174322-174328] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | - mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 - lines Fix something I messed up in the merge I just did ........ - - * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb - 2009) | 20 lines Merged revisions 174282 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb - 2009) | 12 lines Don't do an SRV lookup if a port is specified - RFC 3263 says to do A record lookups on a hostname if a port has - been specified, so that's what we're going to do. See section - 4.2. (closes issue #14419) Reported by: klaus3000 Patches: - patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 - (license 65) ........ ................ - -2009-02-09 14:50 +0000 [r174220] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, - 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 - lines Don't overwrite our pointer to the music class when music - on hold stops. We will use this if it starts again to see if we - can resume the music where it left off. (closes issue #14407) - Reported by: mostyn ........ ................ - -2009-02-07 16:17 +0000 [r174151] Russell Bryant <russell@digium.com> - - * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) - | 10 lines Merged revisions 174148 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) - | 2 lines Fix a race condition that could cause a crash. ........ - ................ - -2009-02-06 23:59 +0000 [r174085] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) - | 13 lines Merged revisions 174082 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) - | 5 lines check ast_strlen_zero() before calling ast_strdupa() in - sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter - didn't actually upload a properly-formed patch, instead a - modified chan_sip.c file was uploaded. I created a patch to - determine the changes, then modified the suggested changes to - create a proper fix. The summary above is a complete description - of the changes. (closes issue #13547) Reported by: tecnoxarxa - Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) - Tested by: tecnoxarxa ........ ................ - ------------------------------------------------------------------------ - -2009-02-06 19:29 +0000 [r173986-174042] Joshua Colp <jcolp@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 - lines Don't subscribe to a mailbox on pseudo channels. It is - futile. This solves an issue where duplicated pseudo channels - would cause a crash because the first one would unsubscribe and - the next one would also try to unsubscribe the same subscription. - (closes issue #14322) Reported by: amessina ........ - - * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | - 15 lines Merged revisions 173967-173968 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 - lines Some clients do not put the call-id for replaces at the - beginning, so support it being anywhere in the string. (closes - issue #14350) Reported by: fhackenberger ........ r173968 | file - | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a - debug message I put in by accident. ........ ................ - -2009-02-06 16:33 +0000 [r173963] Matthew Nicholson <mnicholson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb - 2009) | 14 lines Merged revisions 173917 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb - 2009) | 7 lines Limit the addition of the Contact header in SIP - responses according to various SIP RFCs. (closes issue #13602) - Reported by: hjourdain Tested by: mnicholson ........ - ................ - -2009-02-05 23:51 +0000 [r173774-173777] Mark Michelson <mmichelson@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 173776 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, - 05 Feb 2009) | 14 lines Update extensions.conf.sample to be - correct. In trunk, the only necessary change pointed out was that - the call to ChanIsAvail uses an option that has been removed. For - the 1.6.1 branch, however, it appears that the sample file is - badly in need of updating since there are |'s used all over the - place there. My tentative plan is just to copy trunk's sample - config file to those branches since the info there is most - up-to-date and should be correct for use in 1.6.1 Thanks to macli - in #asterisk-dev for bringing this up ........ - - * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb - 2009) | 7 lines Properly set "seen" and "unseen" flags when - moving messages from the new to the old folder when using IMAP - for voicemail storage (closes issue #13905) Reported by: jaroth - Patches: foldermove_v2.patch uploaded by jaroth (license 50) - ........ - -2009-02-05 21:04 +0000 [r173698] Jeff Peeler <jpeeler@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 - (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) - | 12 lines Add new configuration option to make shared IMAP - mailboxes function as expected. The new option is "imapvmshareid" - which is an ID to tag multiple mailboxes using the same IMAP - storage location to function as one mailbox. This allows all - messages to be retrieved for any user in the group. The patch - alters the 'X-Asterisk-VM-Extension' header that is responsible - for matching voicemails for a given user. (closes issue #13673) - Reported by: howardwilkinson ........ ................ - -2009-02-05 20:34 +0000 [r173590-173694] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 173693 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb - 2009) | 20 lines Merged revisions 173692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb - 2009) | 12 lines Fix situations where queue members could be - autopaused unexpectedly Specifically, this patch prevents us from - autopausing members when we receive a busy or congestion frame - from them. (closes issue #14376) Reported by: fiddur Patches: - 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur - ........ ................ - - * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 - (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb - 2009) | 3 lines Add some missing cleanup to app_mixmonitor - ........ ................ - - * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 - (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb - 2009) | 25 lines Fix a problem where a channel pointer becomes - invalid due to masquerading or hanging up. app_mixmonitor runs - its own thread to monitor the channel's activity and write the - mixed audio to a file. Since this thread runs independently of - the channel, it is possible that the mixmonitor thread's channel - pointer will point to freed memory when the channel either is - masqueraded or hangs up (technically, both cases are hangups, but - we need to handle the cases slightly differently). The solution - for this is to employ a datastore, which has the nice benefit of - allowing us to hook into channel masquerades and hangups and - update our pointer as necessary. If this looks familiar, this - same technique is employed in app_chanspy. app_chanspy is a bit - more involved since it does a lot more operations on the channel - that is being spied upon. app_mixmonitor does have an extra touch - that app_chanspy doesn't have, though. Since there is a thread - race between the channel's thread and the mixmonitor thread on a - hangup, we em- ploy a condition-and-boolean combination to ensure - that the channel thread finishes with our structure before the - mixmonitor thread attempts to free it. No crashes! (closes issue - #14374) Reported by: aragon Patches: 14374.patch uploaded by - putnopvut (license 60) Tested by: aragon, putnopvut ........ - ................ - -2009-02-05 16:23 +0000 [r173554] Jeff Peeler <jpeeler@digium.com> - - * build_tools/menuselect-deps.in: fix WORKING_FORK detection - -2009-02-05 00:11 +0000 [r173548] Tilghman Lesher <tlesher@digium.com> - - * build_tools/menuselect-deps.in: regenerate with bootstrap.sh - -2009-02-04 23:44 +0000 [r173546-173547] Jeff Peeler <jpeeler@digium.com> - - * /: I messed up and accidentally reverted the trunk-merged prop - before committing 173546. Added it manually. - - * main/features.c: Merged revisions 173500 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) - | 23 lines Merged revisions 173211 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) - | 17 lines Parking attempts made to one end of a bridge no longer - will hang up due to a parking failure. Parking attempts made - using either one-touch, or doing either a blind or assisted - transfer to the parking extension now keep up the bridge instead - of hanging up the attempted parked party. Normal causes for the - parking attempt to fail includes the specific specified extension - (via PARKINGEXTEN) not being available or if all the parking - spaces are currently in use. To avoid having to reverse a - masquerade park_space_reserve was made to provide foresight if a - parking attempt will succeed and if so reserve the parking space. - (closes issue #13494) Reported by: mdu113 Reviewed by Russell: - http://reviewboard.digium.com/r/133/ ........ ................ - -2009-02-04 22:23 +0000 [r173534] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 173507 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | - mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 - lines Fix some areas where the incorrect interface was passed to - ast_device_state I swear it feels like I already did this once... - (closes issue #14359) Reported by: francesco_r ........ - -2009-02-04 18:55 +0000 [r173460] Tilghman Lesher <tlesher@digium.com> - - * main/tcptls.c, /: Merged revisions 173458 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | - tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines - When using a socket as a FILE *, the stdio functions will - sometimes try to do an fseek() on the stream, which is an invalid - operation for a socket. Turning off buffering explicitly lets the - stdio functions know they cannot do this, thus avoiding a - potential error. (closes issue #14400) Reported by: fnordian - Patches: tcptls.patch uploaded by fnordian (license 110) ........ - -2009-02-04 17:46 +0000 [r173355-173398] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb - 2009) | 11 lines Merged revisions 173396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb - 2009) | 3 lines Revert my previous change because it was stupid - ........ ................ - - * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb - 2009) | 11 lines Merged revisions 173392 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb - 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever - matter, but it's needed. ........ ................ - - * /, main/file.c: Merged revisions 173354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | - mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 - lines Fix a problem where file playback would cause fds to remain - open forever The problem came from the fact that a frame read - from a format interpreter was not freed. Adding a call to - ast_frfree fixed this. The explanation for why this caused the - problem is a bit complex, but here goes: There was a problem in - all versions of Asterisk where the embedded frame of a filestream - structure was referenced after the filestream was freed. This was - fixed by adding reference counting to the filestream structure. - The refcount would increase every time that a filestream's frame - pointer was pointing to an actual frame of data. When the frame - was freed, the refcount would decrease. Once the refcount reached - 0, the filestream was freed, and as part of the operation, the - open files were closed as well. Thus it becomes more clear why a - missing ast_frfree would cause a reference leak and cause the - files to not be closed. You may ask then if there was a frame - leak before this patch. The answer to that is actually no! The - filestream code was "smart" enough to know that since the frame - we received came from a format interpreter, the frame had no - malloced data and thus didn't need to be freed. Now, however, - there is cleanup that needs to be done when we finish with the - frame, so we do need to call ast_frfree on the frame to be sure - that the refcount for the filestream is decremented - appropriately. (closes issue #14384) Reported by: fiddur Patches: - 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, - putnopvut ........ - -2009-02-04 00:45 +0000 [r173312] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 173311 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | - tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 - lines Ensure that commas placed in the middle of extension - character classes do not interfere with correct parsing of the - extension. Also, if an unterminated character class DOES make its - way into the pbx core (through some other method), ensure that it - does not crash Asterisk. (closes issue #14362) Reported by: - Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by - Corydon76 (license 14) Tested by: Corydon76 ........ - -2009-02-03 23:41 +0000 [r173250] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing - of calls. Fixes issue with IAX2 transfers not taking place. As it - was, a call that was being transfered would never be handed off - correctly to the call ends because of how call numbers were - stored in a hash table. The hash table, "iax_peercallno_pvt", - storing all the current call numbers did not take into account - the complications associated with transferring a call, so a - separate hash table was required. This second hash table - "iax_transfercallno_pvt" handles calls being transfered, once the - call transfer is complete the call is removed from the transfer - hash table and added to the peer hash table resuming normal - operations. Addition functions were created to handle storing, - removing, and comparing items in the iax_transfercallno_pvt - table. (issue #13468) Review: - http://reviewboard.digium.com/r/140/ - -2009-02-03 00:26 +0000 [r173111] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 173104 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 - (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) - | 5 lines Add warning to standard config, that globals may be - overridden by other dialplan configuration files. (closes issue - #14388) Reported by: macli ........ ................ - -2009-02-02 23:59 +0000 [r173068] Terry Wilson <twilson@digium.com> - - * /, main/features.c: Merged revisions 173067 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) - | 9 lines Merged revisions 173066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) - | 2 lines Fix a feature inheritance bug I added after code review - ........ ................ - -2009-02-02 18:15 +0000 [r172896] Leif Madsen <lmadsen@digium.com> - - * /, configs/res_ldap.conf.sample: Merged revisions 172894 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 - Feb 2009) | 7 lines Update the res_ldap.conf file with a better - working example. (closes issue #13861) Reported by: scramatte - Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage - (license 10) Tested by: jcovert ........ - -2009-02-01 02:45 +0000 [r172707-172742] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) - | 4 lines Blank argument crashes Asterisk (closes issue #14377) - Reported by: amorsen ........ - - * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) - | 7 lines Don't increment the loop, now that incrementing is - taken care of by the decoder function. (closes issue #14363) - Reported by: andrew53 Patches: func_strings_filter.patch uploaded - by andrew53 (license 519) ........ - -2009-01-31 00:06 +0000 [r172635-172637] Terry Wilson <twilson@digium.com> - - * configs/features.conf.sample, /: Merged revisions 172581 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 - Jan 2009) | 2 lines Remove incorret line from sample config - ........ - - * configs/features.conf.sample, apps/app_dial.c, - main/global_datastores.c, /, main/features.c, - include/asterisk/global_datastores.h, CHANGES: Merged revisions - 172580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) - | 44 lines Merged revisions 172517 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) - | 37 lines Fix feature inheritance with builtin features When - using builtin features like parking and transfers, the - AST_FEATURE_* flags would not be set correctly for all instances - when either performing a builtin attended transfer, or parking a - call and getting the timeout callback. Also, there was no way on - a per-call basis to specify what features someone should have on - picking up a parked call (since that doesn't involve the Dial() - command). There was a global option for setting whether or not - all users who pickup a parked call should have - AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or - PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan - variable which can be set either in the dialplan or with setvar - in channels that support it. This variable can be set to any - combination of 't', 'k', 'w', and 'h' (case insensitive matching - of the equivalent dial options), to set what features should be - activated on this channel. The patch moves the setting of the - features datastores into the bridging code instead of app_dial to - help facilitate this. 2) adds global options parkedcallparking, - parkedcallhangup, and parkedcallrecording to be similar to the - parkedcalltransfers option for globally setting features. 3) has - builtin_atxfer call builtin_parkcall if being transfered to the - parking extension since tracking everything through multiple - masquerades, etc. is difficult and error-prone 4) attempts to fix - all cases of return calls from parking and completed builtin - transfers not having the correct permissions (closes issue - #14274) Reported by: aragon Patches: - fix_feature_inheritence.diff.txt uploaded by otherwiseguy - (license 396) Tested by: aragon, otherwiseguy Review - http://reviewboard.digium.com/r/138/ ........ ................ - -2009-01-30 22:23 +0000 [r172604] Mark Michelson <mmichelson@digium.com> - - * /, include/asterisk/channel.h: Merged revisions 172598 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, - 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h - ........ - -2009-01-29 23:47 +0000 [r172503] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, apps/app_nbscat.c, /, autoconf/ast_func_fork.m4, - apps/app_festival.c, build_tools/menuselect-deps.in, configure, - apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c, - apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: - Merged revisions 172441 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) - | 16 lines Merged revisions 172438 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) - | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at - startup. Otherwise, if Asterisk runs as a non-root user and the - administrator does a 'restart now', Asterisk loses the ability to - set QOS on packets. (closes issue #14004) Reported by: nemo - Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 - (license 14) Tested by: Corydon76 ........ ................ - -2009-01-29 21:35 +0000 [r172434] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged - revisions 172400 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | - rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 - lines channels/chan_dahdi.c * Added doxygen comments to the major - dahdi structures. * Fixed PRI and SS7 using an incorrect string - value if the extension delimiter is not present in the Dial() - function. * Fixed SS7 not checking if the dialed extension is at - least as long as the stripmsd option. * Fixed PRI not handling - unknown TON/NPI prefix letters correctly. * Fixed some - uninitialized string variables on FXS ports. - configs/chan_dahdi.conf.sample * Updated some documentation. - ........ - -2009-01-29 16:49 +0000 [r172316] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample, /: Merged revisions 172315 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 - Jan 2009) | 2 lines Better document mode=multirow, based upon a - conversation with Jared. ........ - -2009-01-29 13:51 +0000 [r172273] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 - Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a - couple of fields. closes issue #14339) Reported by: fiddur - Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) - ........ - -2009-01-29 09:56 +0000 [r172217] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 - lines Merged revisions 172169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 - lines Make sure that we always add the hangupcause headers. In - some cases, the owner was disconnected before we checked for the - cause. This patch implements a temporary storage in the pvt and - use that instead. The code is based on ideas from code from - Adomjan in issue #13385 (Add support for Reason: header) Thanks - to Klaus Darillion for testing! (closes issue #14294) related to - issue #13385 Reported by: klaus3000 and adomjan Patches: - bug14294b.diff uploaded by oej (license 306) Based on - 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan - (license 487) Tested by: oej, klaus3000 ........ ................ - -2009-01-28 20:41 +0000 [r172065] Steve Murphy <murf@digium.com> - - * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, - main/features.c, include/asterisk/channel.h: Merged revisions - 172063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | - 52 lines Merged revisions 172030 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | - 46 lines This patch fixes h-exten running misbehavior in - manager-redirected situations. What it does: 1. A new Flag value - is defined in include/asterisk/channel.h, - AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the - bridge hangup exten code not to run the h-exten there (nor - publish the bridge cdr there). It will done at the pbx-loop level - instead. 2. In the manager Redirect code, I set this flag on the - channel if the channel has a non-null pbx pointer. I did the same - for the second (chan2) channel, which gets run if name2 is set... - and the first succeeds. 3. I restored the ending of the cdr for - the pbx loop h-exten running code. Don't know why it was removed - in the first place. 4. The first attempt at the fix for this bug - was to place code directly in the async_goto routine, which was - called from a large number of places, and could affect a large - number of cases, so I tested that fix against a fair number of - transfer scenarios, both with and without the patch. In the - process, I saw that putting the fix in async_goto seemed not to - affect any of the blind or attended scenarios, but still, I was - was highly concerned that some other scenarios I had not tested - might be negatively impacted, so I refined the patch to its - current scope, and jmls tested both. In the process, tho, I saw - that blind xfers in one situation, when the one-touch blind-xfer - feature is used by the peer, we got strange h-exten behavior. So, - I inserted code to swap CDRs and to set the HANGUP_DONT field, to - get uniform behavior. 5. I added code to the bridge to obey the - HANGUP_DONT flag, skipping both publishing the bridge CDR, and - running the h-exten; they will be done at the pbx-loop (higher) - level instead. 6. I removed all the debug logs from the patch - before committing. 7. I moved the AUTOLOOP set/reset in the - h-exten code in res_features so it's only done if the h-exten is - going to be run. A very minor performance improvement, but - technically correct. (closes issue #14241) Reported by: jmls - Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer - uploaded by murf (license 17) Tested by: murf, jmls ........ - ................ - -2009-01-28 17:28 +0000 [r171965] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 - (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 - Jan 2009) | 2 lines Clarify log message (suggested by manxpower - on #asterisk-dev) ........ ................ - -2009-01-28 13:18 +0000 [r171846] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Merged revisions 171838 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, - 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 - lines Add a better explanation of the difference between the - device namespace and the dialplan for newbies. ........ - ................ - -2009-01-27 22:00 +0000 [r171619-171692] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 - (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan - 2009) | 39 lines Fix devicestate problems for "always-on" agent - channels A revision to chan_agent attempted to "inherit" the - device state of the underlying channel in order to report the - device state of an agent channel more accurately. The problem - with the logic here is that it makes no sense to use this for - always-on agents. If the agent is logged in, then to the - underlying channel, the agent will always appear to be "in use," - no matter if the agent is on a call or not. The reason is that to - the underlying channel, the channel is currently in use on a call - to the AgentLogin application. The most common cause that I found - for this issue to occur was for a SIP channel to be the - underlying channel type for an Agent channel. If the SIP phone - re-registers, then the registration will cause the device state - core to query the device state of the SIP channel. Since the SIP - channel is in use, the Agent channel would also inherit this - status. Once the agent channel was set to "in use" there was no - way that the device state could change on that channel unless the - agent logged out. The solution for this problem is a bit - different in 1.4 than it is in the other branches. In 1.4, there - will be a one-line fix to make sure that only callback agents - will inherit device state from their underlying channel type. For - the other branches of Asterisk, since callback support has been - removed, there is also no need for device state inheritance in - chan_agent, so I will simply be removing it from the code. In - addition, the 1.4 source is getting a new comment to help the - next person who edits chan_agent.c. I'm adding a comment that a - agent_pvt's loginchan field may be used to determine if the agent - is a callback agent or not. (closes issue #14173) Reported by: - nathan Patches: 14173.patch uploaded by putnopvut (license 60) - Tested by: nathan, aramirez ........ ................ - - * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan - 2009) | 26 lines Merged revisions 171621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan - 2009) | 18 lines Prevent a crash from occurring when a jitter - buffer interpolated frame is removed from a slinfactory - slinfactory used the "samples" field of an ast_frame in order to - determine the amount of data contained within the frame. In - certain cases, such as jitter buffer interpolated frames, the - frame would have a non-zero value for "samples" but have NULL - "data" This caused a problem when a memcpy call in - ast_slinfactory_read would attempt to access invalid memory. The - solution in use here is to never feed frames into the slinfactory - if they have NULL "data" (closes issue #13116) Reported by: - aragon Patches: 13116.diff uploaded by putnopvut (license 60) - ........ ................ - - * /, apps/app_queue.c: Merged revisions 171618 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | - mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 - lines Fix queue crashes that would occur after the calling - channel was masqueraded. The data passed to the - end_bridge_callback was assumed to be data which was still - stack'd. The problem was that with some call features, attended - transfers in particular, a new bridge thread is started once the - feature completes, meaning that when the end_bridge_callback is - called, the end_bridge_callback_data was invalid. To fix this - problem, there are two measures taken 1. Instead of pointing to - stacked data, we now used heap-allocated data for passing to the - end_bridge_callback in app_queue 2. Since bridges can end - multiple times on a single logical call, we wait until the final - bridge is broken to actually set any queue variables. This is - accomplished through reference-counting and the use of an - end_bridge_callback_data_fixup function in app_queue.c (closes - issue #14260) Reported by: ccesario Patches: 14260.patch uploaded - by putnopvut (license 60) Tested by: ccesario ........ - -2009-01-27 16:15 +0000 [r171594-171595] Matthew Fredrickson <creslin@digium.com> - - * main/ast_expr2.c, main/ast_expr2.h: Revert some changes that - shouldn't have made it in - - * main/ast_expr2.c, channels/chan_dahdi.c, main/ast_expr2.h: Make - sure we do not go into alarm on PTMP links with non persistent - D-channels - -2009-01-27 15:13 +0000 [r171529] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 - lines Solving the same issue, but a bit different in trunk... - Merged revisions 171527 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 - lines Use the same branch tag in CANCEL as in INVITE Originally - putnopvut implemented some changes in revision 142079 that - according to the bug report seemed to have worked then, but - somehow fails now. I guess code, as humans, get old and forget - stuff. Anyway, this bug caused CANCEL not to work with picky - systems. Thanks Fredrik for pointing out where the bug in the SIP - messaging was. (closes issue #14346) Reported by: oej Patches: - bug14346.diff uploaded by oej (license 306) Tested by: oej - ........ ................ - -2009-01-26 14:02 +0000 [r171327] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | - 17 lines Merged revisions 171264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 - lines Don't retransmit 401 on REGISTER requests when - alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 - Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by - klaus3000 (license 65) Tested by: klaus3000 ........ - ................ - -2009-01-26 00:03 +0000 [r171189] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) - | 13 lines Merged revisions 171187 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) - | 6 lines Correctly track the hookstate (closes issue #13686) - Reported by: itiliti Patches: 20081013__bug13686.diff.txt - uploaded by Corydon76 (license 14) ........ ................ - -2009-01-25 13:38 +0000 [r170981] Sean Bright <sean.bright@gmail.com> - - * /, apps/app_page.c: Merged revisions 170980 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan - 2009) | 16 lines Merged revisions 170979 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan - 2009) | 9 lines Resolve a logic error that was causing Page() to - crash when more than one channel was specified. (closes issue - #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt - uploaded by seanbright (license 71) Tested by: kc0bvu ........ - ................ - -2009-01-25 02:50 +0000 [r170944] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) - | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second - part of this macro is written as 0[a] instead of a[0], it will - force a failure if the macro is used on a C++ object that - overloads the [] operator. ........ - -2009-01-24 13:56 +0000 [r170838] Tilghman Lesher <tlesher@digium.com> - - * configs/res_odbc.conf.sample, /: Merged revisions 170837 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 - (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 - Jan 2009) | 2 lines Remove superfluous implementation note - (closes issue #14319) ........ ................ - -2009-01-23 23:52 +0000 [r170830] Richard Mudgett <rmudgett@digium.com> - - * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | - rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line - Fix asterisk.pdf generation if branch name has an underscore in - it. ........ - -2009-01-23 22:59 +0000 [r170791] Russell Bryant <russell@digium.com> - - * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | - russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines - Don't blow up if a branch name has an underscore in it ........ - -2009-01-23 20:56 +0000 [r170685-170721] Mark Michelson <mmichelson@digium.com> - - * configs/res_odbc.conf.sample, /: Merged revisions 170720 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 - (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan - 2009) | 8 lines Add notes to the idlecheck explanation in - res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 - Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by - klaus3000 (license 65) ........ ................ - - * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 - (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan - 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use - deprecated syntax * Convert Wait,1 to Wait(1) * Convert - SetLanguage to Set(CHANNEL(language)) * Use 'n' for all - priorities beyond the first Also added test for Chinese numbers, - too. (closes issue #14320) Reported by: dant Patches: - i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license - 670) ........ ................ - -2009-01-23 20:19 +0000 [r170659] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 170652 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | - 11 lines Merged revisions 170648 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 - lines When a channel is answered make sure any indications - currently playing stop. Usually the phone would do this but if - the channel was already answered then they are being generated by - Asterisk and we darn well need to stop them. (closes issue - #14249) Reported by: RadicAlish ........ ................ - -2009-01-23 Tilghman Lesher <tlesher@digium.com> - - * Released 1.6.0.5 - - * channels/chan_iax2.c: Regression fixes for security fix AST-2009-001 - -2009-01-06 Tilghman Lesher <tlesher@digium.com> - - * Released 1.6.0.3 - - * channels/chan_iax2.c: Security fix AST-2009-001 - -2008-12-03 Tilghman Lesher <tlesher@digium.com> - - * Released 1.6.0.3-rc1 - -2008-12-03 14:13 +0000 [r160482] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) - | 14 lines Merged revisions 160480 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) - | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I - guess that having only ip-phones in mind is not a good approach. - Since it is possible to have a sip proxy connected to asterisk we - could receive a 407 (unauthorized) or 483 (too many hops) as - response and dialog ending would not be a good behavior." So - modified. ........ ................ - -2008-12-03 00:53 +0000 [r160427] Sean Bright <sean.bright@gmail.com> - - * Makefile: Fix some 'make menuselect' breakage introduced by - recent merges. - -2008-12-02 23:22 +0000 [r160386-160393] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) - | 12 lines Merged revisions 156386 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) - | 5 lines When using call limits under 1 second, infinite call - lengths are allowed, instead. (closes issue #13851) Reported by: - ruddy ........ ................ - - * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) - | 11 lines Merged revisions 156289 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) - | 3 lines For whatever reason, gcc only warned me about the - possible use of an uninitialized variable when compiling 1.6.1. - ........ ................ - - * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) - | 16 lines Merged revisions 156178 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) - | 8 lines (closes issue #13173) Reported by: pep This change adds - an announce_thread responsible for playing announcements to an - existing conference. This allows all announcing to be immediately - stopped if necessary but more importantly allows other threads - that need to play something to not block. There are multiple - benefits to this, but the actual bug is for solving the scenario - for a channel to be unusable after hang up for the entire - duration of the parting announcement. The parting announcement - can be extremely long depending on what the user recorded upon - joining the conference. Reviewed by Russell on Review Board: - http://reviewboard.digium.com/r/25/ ........ ................ - - * main/astobj2.c, main/asterisk.c, apps/app_while.c, - apps/app_dial.c, main/pbx.c, channels/chan_misdn.c, - main/manager.c, /, apps/app_meetme.c, channels/chan_sip.c, - channels/chan_skinny.c, include/asterisk/astobj2.h, - channels/chan_agent.c, channels/chan_h323.c, - channels/chan_iax2.c: Merged revisions - 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500 - (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) - | 3 lines Cannot join detached threads. See - http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html - (Closes issue #13400) ........ ................ r153122 | - tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 - lines Merged revisions 153114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) - | 3 lines Turn off qualify on uncached realtime peers. (Closes - issue #13383) ........ ................ r154264 | tilghman | - 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded - merge of revisions 154263 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) - | 3 lines Make the monitor thread non-detached, so it can be - joined (suggested by Russell on -dev list). ........ - ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 - (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) - | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state - when it receives the indication AST_CONTROL_RINGING. ........ - ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 - (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) - | 9 lines On busy systems, it's possible for the values checked - within a single line of code to change, unless the structure is - locked to ensure a consistent state. (closes issue #13717) - Reported by: kowalma Patches: 20081102__bug13717.diff.txt - uploaded by Corydon76 (license 14) Tested by: kowalma ........ - ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 - (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) - | 7 lines Clarify error message. (closes issue #13809) Reported - by: denke Patches: 20081104__bug13809.diff.txt uploaded by - Corydon76 (license 14) Tested by: denke ........ ................ - r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov - 2008) | 22 lines Merged revisions 155861 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov - 2008) | 14 lines Channel drivers assume that when their indicate - callback is invoked, that the channel on which the callback was - called is locked. This patch corrects an instance in chan_agent - where a channel's indicate callback is called directly without - first locking the channel. This was leading to some observed - locking issues in chan_local, but considering that all channel - drivers operate under the same expectations, the generic fix in - chan_agent is the right way to go. AST-126 ........ - ................ r156166 | russell | 2008-11-12 11:38:20 -0600 - (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) - | 7 lines Move the sanity check that makes sure "always fork" is - not set along with the console option to be after the code that - reads options from asterisk.conf. This resolves a situation where - Asterisk can start taking up 100% when misconfigured. (Thanks to - Bryce Porter (x86 on IRC) for letting me log in to his system to - figure out what was causing the 100% CPU problem.) ........ - ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 - (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) - | 6 lines If the SLA thread is not started, then reload causes a - memory leak. (closes issue #13889) Reported by: eliel Patches: - app_meetme.c.patch uploaded by eliel (license 64) ........ - ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 - (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) - | 7 lines Provide more space for all the data which can appear in - an originating channel name. (closes issue #13398) Reported by: - bamby Patches: manager.c.diff uploaded by bamby (license 430) - ........ ................ r156756 | tilghman | 2008-11-13 - 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions - 156755 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) - | 6 lines ast_waitfordigit() requires that the channel be up, for - no good logical reason. This prevents While/EndWhile from working - within the "h" extension. Reported by: jgalarneau (for ABE C.2) - Fixed by: me ........ ................ r158066 | mmichelson | - 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged - revisions 158053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov - 2008) | 12 lines Make sure to set the hangup cause on the calling - channel in the case that ast_call() fails. For incoming SIP - channels, this was causing us to send a 603 instead of a 486 when - the call-limit was reached on the destination channel. (closes - issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded - by putnopvut (license 60) Tested by: blitzrage ........ - ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 - (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov - 2008) | 16 lines We don't handle 4XX responses to BYE well. - According to section 15 of RFC 3261, we should terminate a dialog - if we receive a 481 or 408 in response to our BYE. Since I am - aware of at least one phone manufacturer who may sometimes send a - 404 as well, I am being liberal and saying that any 4XX response - to a BYE should result in a terminated dialog. (closes issue - #12994) Reported by: pabelanger Patches: 12994.patch uploaded by - putnopvut (license 60) Closes AST-129 ........ ................ - r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) - | 10 lines Merged revisions 158539 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) - | 2 lines When compiling with DEBUG_THREADS, report the real - file/func/line for ao2_lock/ao2_unlock ........ ................ - r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) - | 12 lines Merged revisions 158600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) - | 5 lines The passed extension may not be the same in the list as - the current entry, because we strip spaces when copying the - extension into the structure. Therefore, use the copied item to - place the item into the list. (found by lmadsen on -dev, fixed by - me) ........ ................ r159276 | tilghman | 2008-11-25 - 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions - 159269 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) - | 7 lines Don't try to send a response on a NULL pvt. (closes - issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch - uploaded by eliel (license 64) Tested by: barthpbx ........ - ................ - - * configs/features.conf.sample, apps/app_voicemail.c, - apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, /, - channels/chan_sip.c, apps/app_queue.c: Merged revisions - 152216,152287,152369,152467,152569,152605 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) - | 13 lines Merged revisions 152215 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) - | 6 lines Inherit ALL elements of CallerID across a local - channel. (closes issue #13368) Reported by: Peter Schlaile - Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 - (license 14) ........ ................ r152287 | jpeeler | - 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged - revisions 152286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) - | 2 lines Buffer policy setting for half is not needed. ........ - ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500 - (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) - | 8 lines Reset all DIAL variables back to blank, in case Dial is - called multiple times per call (which could otherwise lead to - inconsistent status reports). (closes issue #13216) Reported by: - ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 - (license 14) Tested by: ruddy ........ ................ r152467 | - tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 - lines Merged revisions 152463 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) - | 3 lines Quoting in the wrong direction (Fixes AST-107) ........ - ................ r152569 | russell | 2008-10-29 00:34:26 -0500 - (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) - | 7 lines Fix an incorrect usage of sizeof() (closes issue - #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch - uploaded by andrew53 (license 519) ........ ................ - r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | - 22 lines Merged revisions 152538 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | - 14 lines A little documentation cross-ref between features and - dial and queue... I wasted some time (stupidly) trying to get the - one-touch parking stuff working, because it didn't occur to me - that I had to also have the corresponding options in the dial - command! Duh! (In all this time, I never set this up before!) So, - to keep some poor fool from suffering the same fate, I made the - features.conf.sample file mention the corresponding opts in - dial/queue; and the docs for dial/app specifically mention the - corresponding decls in the feature.conf file. I hope this doesn't - spoil some vast, eternal plan... ........ ................ - - * apps/app_speech_utils.c, apps/app_voicemail.c, Makefile, - channels/chan_dahdi.c, /, channels/chan_sip.c, - include/asterisk/audiohook.h, apps/app_waitforsilence.c, - main/features.c, main/audiohook.c, apps/app_queue.c: Merged - revisions - 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, - 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 - lines If we receive DTMF make sure that the state of the speech - structure goes back to being not ready. (issue #LUMENVOX-8) - ........ ................ r147689 | kpfleming | 2008-10-08 - 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions - 147681 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct - 2008) | 3 lines when parsing a text configuration option, ensure - that the buffer on the stack is actually large enough to hold the - legal values of that option, and also ensure that sscanf() knows - to stop parsing if it would overrun the buffer (without these - changes, specifying "buffers=...,immediate" would overflow the - buffer on the stack, and could not have worked as expected) - ........ ................ r148000 | tilghman | 2008-10-09 - 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions - 147997 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) - | 4 lines When blank, callerid name and number should display - "unknown caller" in voicemail emails. (Closes issue #13643) - ........ ................ r148112 | mmichelson | 2008-10-09 - 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions - 146026 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | - 18 lines (closes issue #13579) Reported by: dwagner (closes issue - #13584) Reported by: dwagner Tested by: murf, putnopvut The - thought occurred to me that the res= from the extension spawn was - ending up being returned from the bridge. "Thou shalt not poison - the return value". Made the change and it appears to allow blind - xfers to work as normal. If I'm wrong, reopen the bugs. But it - looks good to me! Many thanks to putnopvut for helping me - reproduce this! ........ ................ r148268 | tilghman | - 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged - revisions 148257 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) - | 7 lines User not notified of temporary greeting, if ODBC - storage is in use. (closes issue #13659) Reported by: moliveras - Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 - (license 14) Tested by: moliveras ........ ................ - r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) - | 11 lines Merged revisions 148916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) - | 4 lines Ensure that mail headers are 7-bit clean, even when - UTF-8 characters are used in headers like 'Subject' and 'To'. - Closes AST-107. ........ ................ r148988 | tilghman | - 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged - revisions 148987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) - | 2 lines Some compilers warn, some don't. Fixing. ........ - ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 - (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) - | 6 lines Check correct values in the return of ast_waitfor(); - also, get rid of a possible memory leak. (closes issue #13658) - Reported by: explidous Patch by: me ........ ................ - r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct - 2008) | 15 lines Merged revisions 149130 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct - 2008) | 7 lines Don't allow reserved characters to be used in - register lines in sip.conf. (closes issue #13570) Reported by: - putnopvut ........ ................ r149201 | mmichelson | - 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged - revisions 149200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct - 2008) | 12 lines Update the queue with the correct number of - calls and whether the call was completed within the service level - when a transfer takes place. This way, we do not "break" the - leastrecent and fewestcalls strategies by not logging a call - until after the transferred call has ended. (closes issue #13395) - Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded - by Marquis (license 32) ........ ................ r149205 | - mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 - lines Merged revisions 149204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct - 2008) | 12 lines Add a tolerance period for sync-triggered - audiohooks so that if packetization of audio is close (but not - equal) we don't end up flushing the audiohooks over small - inconsistencies in synchronization. Related to issue #13005, and - solves the issue for most people who were experiencing the - problem. However, a small number of people are still experiencing - the problem on long calls, so I am not closing the issue yet - ........ ................ r149208 | mmichelson | 2008-10-14 - 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions - 149207 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct - 2008) | 9 lines Call register_peer_exten even in the case that - the peer's IP/port does not change. (closes issue #13309) - Reported by: dimas Patches: v2-13309.patch uploaded by dimas - (license 88) ........ ................ - - * channels/misdn/isdn_lib.c, Makefile, channels/chan_dahdi.c, - channels/chan_misdn.c, main/manager.c, /: Merged revisions - 115313,121770,123272,139624,140205,144257 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008) - | 10 lines Merged revisions 115312 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008) - | 2 lines Reverse order, such that user configs override default - selections ........ ................ r121770 | crichter | - 2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines Merged - revisions 121751 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) - | 1 line fixed issue with previous commit, the find_free_channel - test for channels which where inuse was broken. ........ - ................ r123272 | russell | 2008-06-17 10:52:13 -0500 - (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) - | 4 lines Fix a memory leak in astobj2 that was pointed out by - seanbright. When a container got destroyed, the underlying bucket - list entry for each object that was in the container at that time - did not get free'd. ........ ................ r139624 | jpeeler | - 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines Merged - revisions 139621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) - | 5 lines (closes issue #13359) Reported by: Laureano Patches: - originate_channel_check.patch uploaded by Laureano (license 265) - ........ ................ r140205 | jpeeler | 2008-08-26 13:48:55 - -0500 (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) - | 9 lines (closes issue #12071) Reported by: tzafrir Patches: - dahdi_close.diff uploaded by tzafrir (license 46) Tested by: - tzafrir, jpeeler This patch fixes closing open file descriptors - in the case of an error. ........ ................ r144257 | - crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines - Merged revisions 144238 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) - | 1 line improved helptext of misdn_set_opt. ........ - ................ - -2008-12-02 18:05 +0000 [r160326-160337] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) - | 1 line remove duplicate comment that I accidentally merged - ........ - - * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) - | 7 lines (closes issue #13786) Reported by: tzafrir Readding - DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which - fixes not being able to make outgoing calls on some FXO adapters: - http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 - ........ - -2008-12-02 18:01 +0000 [r160228-160322] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) - | 17 lines Merged revisions 160297 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) - | 10 lines When the text does not match exactly (e.g. RTP/SAVP), - then the %n conversion fails, and the resulting integer is - garbage. Thus, we must initialize the integer and check it - afterwards for success. (closes issue #14000) Reported by: folke - Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke - (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by - folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff - uploaded by folke (license 626) ........ ................ - - * include/asterisk/stringfields.h, apps/app_voicemail.c, - main/cli.c, main/pbx.c, main/frame.c, /, - channels/chan_features.c: Merged revisions 160208 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 - (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) - | 3 lines Ensure that Asterisk builds with --enable-dev-mode, - even on the latest gcc and glibc. ........ ................ - -2008-12-01 23:41 +0000 [r160173] Sean Bright <sean.bright@gmail.com> - - * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged - revisions 160170-160172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec - 2008) | 1 line Pay attention to the return value of system(), - even if we basically ignore it. ................ r160171 | - seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 - line Silence a build warning. (chan_phone.c:810: warning: value - computed is not used) ................ r160172 | seanbright | - 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged - revisions 159976 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) - | 3 lines Get rid of the useless format string and argument in - the Bogus/ manager channelname. Noted by kpfleming and name - Bogus/manager suggested by eliel ........ ................ - -2008-12-01 Tilghman Lesher <tlesher@digium.com> - - * Released 1.6.0.2 - -2008-12-01 21:45 +0000 [r160100] Tilghman Lesher <tlesher@digium.com> - - * /, configure, configure.ac: Merged revisions 160097 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) - | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or - bad things happen. ........ - -2008-12-01 21:07 +0000 [r160096] Sean Bright <sean.bright@gmail.com> - - * include/asterisk.h, /: Merged revisions 154919 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | - seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 - lines Fix a problem found while building res_snmp. ........ - -2008-12-01 17:39 +0000 [r160005] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 160004 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r160004 | russell | 2008-12-01 11:34:31 -0600 - (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) - | 6 lines Apply some logic used in iax2_indicate() to - iax2_setoption(), as well, since they both have the potential to - send control frames in the middle of call setup. We have to wait - until we have received a message back from the remote end before - we try to send any more frames. Otherwise, the remote end will - consider it invalid, and we'll get stuck in an INVAL/VNAK storm. - ........ ................ - -2008-12-01 16:04 +0000 [r159974] Michiel van Baak <michiel@vanbaak.info> - - * main/manager.c, /: Merged revisions 159898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) - | 11 lines Merged revisions 159897 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) - | 4 lines make manager compile on OpenBSD. The last (10th) - argument to ast_channel_alloc here should be a pointer and NULL - is not really a pointer. ........ ................ - -2008-12-01 14:56 +0000 [r159915] Russell Bryant <russell@digium.com> - - * .cleancount, /: Merged revisions 159911 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) - | 10 lines Merged revisions 159900 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) - | 2 lines Force a "make clean" to avoid a bizarre build issue ... - ........ ................ - -2008-11-29 18:37 +0000 [r159855] Kevin P. Fleming <kpfleming@digium.com> - - * utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, Makefile, - include/asterisk/logger.h, include/asterisk/res_odbc.h, - main/srv.c, channels/chan_misdn.c, - include/asterisk/linkedlists.h, main/event.c, - include/asterisk/strings.h, utils/extconf.c, makeopts.in, - include/asterisk/stringfields.h, utils/check_expr.c, - channels/chan_vpb.cc, /, main/utils.c, res/res_config_sqlite.c, - utils/frame.c, channels/misdn_config.c, include/asterisk/astmm.h, - include/asterisk/compat.h, configure, channels/misdn/ie.c, - include/asterisk/module.h, main/features.c, main/dns.c, - funcs/Makefile, include/asterisk/devicestate.h, - include/asterisk/utils.h, channels/chan_sip.c, main/Makefile, - include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac, - channels/chan_agent.c, utils/astman.c, include/asterisk/cli.h, - include/asterisk/channel.h, include/jitterbuf.h, - include/asterisk/manager.h: Merged revisions 159818 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov - 2008) | 18 lines incorporates r159808 from branches/1.4: - ------------------------------------------------------------------------ - r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov - 2008) | 7 lines update dev-mode compiler flags to match the ones - used by default on Ubuntu Intrepid, so all developers will see - the same warnings and errors since this branch already had some - printf format attributes, enable checking for them and tag - functions that didn't have them format attributes in a consistent - way - ------------------------------------------------------------------------ - in addition: move some format attributes from main/utils.c to the - header files they belong in, and fix up references to the - relevant functions based on new compiler warnings ........ - -2008-11-26 19:58 +0000 [r159558] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | - mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 - lines Add some necessary hangup commands in the case that - forwarding a call fails 1) Hang up the original destination if - the local channel cannot be requested. 2) Hang up the local - channel (in addition to the original destination) if ast_call - fails when calling the newly created local channel. This prevents - channels from sticking around forever in the case of a botched - call forward (e.g. to an extension which does not exist). (closes - issue #13764) Reported by: davidw Patches: 13764_v2.patch - uploaded by putnopvut (license 60) Tested by: putnopvut, davidw - ........ - -2008-11-26 19:18 +0000 [r159536] Kevin P. Fleming <kpfleming@digium.com> - - * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, - Makefile.rules: Merged revisions 159534 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov - 2008) | 11 lines Merged revisions 159476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov - 2008) | 7 lines simplify (and slightly bug-fix) the recent - developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' - removes dependency files for .i files that are created in - COMPILE_DOUBLE mode ........ ................ - -2008-11-26 18:40 +0000 [r159478] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, /: Merged revisions 159475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | - tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines - If the config file does not exist, then the first use crashes - Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: - udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage - ........ - -2008-11-26 15:01 +0000 [r159439] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Merged revisions 159437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 | - mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 - lines Don't allow for configuration options to overwrite options - set via channel variables on a reload. (closes issue #13921) - Reported by: davidw Patches: 13921.patch uploaded by putnopvut - (license 60) Tested by: davidw ........ - -2008-11-25 23:09 +0000 [r159374] Steve Murphy <murf@digium.com> - - * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159360 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, - 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | - 15 lines (closes issue #12694) Reported by: yraber Patches: - 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, - laurav Thanks to file (Joshua Colp) for his IAX fix. the change - to cdr.c allows no-answer to percolate up into CDR's, and feels - like the right place to locate this fix; if BUSY is done here, - no-answer should be, too. ........ ................ - -2008-11-25 22:28 +0000 [r159314] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: I don't care what anyone says, this change is - going into 1.6.0. Otherwise, the simple act of logging an agent - in spams the CLI with warning messages about failed reads of the - alertpipe. - -2008-11-25 21:43 +0000 [r159248] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 159247 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 - (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 - (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) - | 7 lines Regression fix for last security fix. Set the iseqno - correctly. (closes issue #13918) Reported by: ffloimair Patches: - 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) - Tested by: ffloimair ........ ................ ................ - -2008-11-25 16:21 +0000 [r159024-159094] Terry Wilson <twilson@digium.com> - - * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | - twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines - Add missing variable declaration for PPC code ........ - - * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) - | 2 lines Make chan_usbradio compile under dev mode ........ - -2008-11-21 22:40 +0000 [r158545] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 158484 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | - 19 lines Merged revisions 158483 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | - 11 lines (closes issue #13871) Reported by: mdu113 This one is - totally my fault. The code doesn't even create a bridge CDR if - the channel CDR has POST_DISABLED. I didn't check for that at the - end of the bridge. Fixed with a few small insertions. Tested. - Looks good. No cdr generated, no crash, no unnecc. data objects - created either. ........ ................ - -2008-11-21 22:13 +0000 [r158542] Russell Bryant <russell@digium.com> - - * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions - 158540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) - | 10 lines Merged revisions 158539 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) - | 2 lines When compiling with DEBUG_THREADS, report the real - file/func/line for ao2_lock/ao2_unlock ........ ................ - -2008-11-21 20:44 +0000 [r158451] Kevin P. Fleming <kpfleming@digium.com> - - * /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, - UPGRADE-1.6.txt, CHANGES: Merged revisions 158449 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov - 2008) | 3 lines as suggested by jtodd, document the purposes of - the CHANGES and UPGRADE files ........ - -2008-11-21 17:12 +0000 [r158376] Terry Wilson <twilson@digium.com> - - * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | - twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines - Reloading the config and having no changes still initialized some - settings to 0. Initialize settings after doing all of the cfg - checks. (closes issue #13942) Reported by: davidw Patches: - cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: - davidw ........ - -2008-11-21 01:23 +0000 [r158231-158267] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 158265-158266 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, - 20 Nov 2008) | 4 lines Use some magic constants to get the right - size for this sscanf statement. Thanks Richard! ........ r158266 - | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 - lines Use a more expressive constant for a 64-bit scanned int - ........ - - * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | - mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 - lines Fix the build for 32-bit systems. %lu is only 32-bits on - 32-bit systems, so we need to use %llu instead. Of course %llu is - 128-bits on 64-bit systems, so we have to cast to unsigned long - long. No harm, but it's sure annoying. ........ - - * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | - mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 - lines Change the remote user agent session version variable from - an int to a uint64_t. This prevents potential comparison problems - from happening if the version string exceeds INT_MAX. This was an - apparent problem for one user who could not properly place a call - on hold since the version in the SDP of the re-INVITE to place - the call on hold greatly exceeded INT_MAX. This also aligns with - RFC 2327 better since it recommends using an NTP timestamp for - the version (which is a 64-bit number). (closes issue #13531) - Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut - (license 60) Tested by: sgofferj ........ - -2008-11-20 19:42 +0000 [r158190] Sean Bright <sean.bright@gmail.com> - - * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | - seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 - lines Fix one case where the application argument was not - converted from a pipe to a comma. This was causing problems with - switch statements with empty expressions. (closes issue #13901) - Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by - seanbright (license 71) Tested by: seanbright Reviewed by: murf - ........ - -2008-11-20 00:12 +0000 [r157738-157976] Kevin P. Fleming <kpfleming@digium.com> - - * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, - channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, - channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, - codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, - channels/misdn, main/db1-ast/mpool, Makefile.rules, res/snmp, - pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 - (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov - 2008) | 7 lines the gcc optimizer frequently finds broken code - (use of uninitalized variables, unreachable code, etc.), which is - good. however, developers usually compile with the optimizer - turned off, because if they need to debug the resulting code, - optimized code makes that process very difficult. this means that - we get code changes committed that weren't adequately checked - over for these sorts of problems. with this build system change, - if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is - turned on, when a source file is compiled it will actually be - preprocessed (into a .i or .ii file), then compiled once with - optimization (with the result sent to /dev/null) and again - without optimization (but only if the first compile succeeded, of - course). while making these changes, i did some cleanup work in - Makefile.rules to move commonly-used combinations of flag - variables into their own variables, to make the file easier to - read and maintain ........ ................ - - * /, res/res_agi.c: Merged revisions 157743 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | - kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line - correct small bug introduced during API conversion ........ - - * apps/app_stack.c, include/asterisk/agi.h, /, channels/chan_sip.c, - res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged - revisions 157706 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | - kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 - lines make some corrections to the ast_agi_register_multiple(), - ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to - be consistent with API guidelines also, move UPGRADE.txt to - UPGRADE-1.6.txt and make the new UPGRADE.txt contain information - about upgrading between Asterisk 1.6 releases ........ - -2008-11-19 00:33 +0000 [r157601] Sean Bright <sean.bright@gmail.com> - - * Makefile, /, build_tools/make_version, configure, configure.ac, - build_tools/make_buildopts_h, makeopts.in: Merged revisions - 157600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | - seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 - lines Fix a few build problems on Solaris (and check for an md5 - utility in configure instead of the icky loop I was doing - before). (closes issue #13842) Reported by: snuffy Patches: - bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff - uploaded by seanbright (license 71) Tested by: snuffy ........ - -2008-11-18 22:59 +0000 [r157307-157541] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Merged revisions 157512 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov - 2008) | 21 lines Merged revisions 157503 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov - 2008) | 13 lines Add some missing invite state changes necessary - in the sip_write function. Not setting the invite state correctly - on the call was resulting in the Record application leaving empty - files. I also have updated the doxygen comment next to the - declaration of the INV_EARLY_MEDIA constant to reflect that we - also use this state when we *send* a 18X response to an INVITE. - (closes issue #13878) Reported by: nahuelgreco Patches: - sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco - (license 162) Tested by: putnopvut ........ ................ - - * channels/chan_sip.c: Once again, Russell to the rescue. Use the - builtin astobj1 lock of the sip_peer and sip_user instead of - adding a new one - - * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | - mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 - lines Based on Russell's advice on the asterisk-dev list, I have - changed from using a global lock in update_call_counter to using - the locks within the sip_pvt and sip_peer structures instead. - ........ - - * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | - mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 - lines * Add a lock to be used in the update_call_counter - function. * Revert logic to mirror 1.4's in the sense that it - will not allow the call counter to dip below 0. These two - measures prevent potential races that could cause a SIP peer to - appear to be busy forever. (closes issue #13668) Reported by: mjc - Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic - (license 586) ........ - - * apps/app_dial.c, channels/chan_local.c, /, main/features.c, - include/asterisk/channel.h, apps/app_followme.c: Merged revisions - 157306 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov - 2008) | 20 lines Merged revisions 157305 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov - 2008) | 12 lines Fix a crash in the end_bridge_callback of - app_dial and app_followme which would occur at the end of an - attended transfer. The error occurred because we initially stored - a pointer to an ast_channel which then was hung up due to a - masquerade. This commit adds a "fixup" callback to the - bridge_config structure to allow for end_bridge_callback_data to - be changed in the case that a new channel pointer is needed for - the end_bridge_callback. ........ ................ - -2008-11-18 18:10 +0000 [r157303] Steve Murphy <murf@digium.com> - - * main/config.c, /: Merged revisions 157302 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | - murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines - (closes issue #13420) Reported by: alex70 Patches: - 13420.13539.patch uploaded by murf (license 17) Tested by: murf, - awk This fixes two problems: a spurious linefeed insertion - probably left over from pre-precomment times. Only generated when - category had no previous comments. The other problem: Insertions - could get the line-numbering out of whack and generate negative - line numbers, causing chunks of line numbers to be emitted, on - the scale of the number of lines up to that point in the file. In - such cases, abort the looping, and all is well. ........ - -2008-11-15 19:47 +0000 [r157107-157165] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged - revisions 157164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov - 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov - 2008) | 1 line dist-clean should remove dependency information - files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 - +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory - dist-clean is run, run clean in that directory first, and when - running top-level dist-clean, do not run subdirectory clean - operations twice ........ ................ - - * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 - Nov 2008) | 13 lines major update to doxygen configuration file: - 1) update to doxygen 1.5.x style file, as used in trunk 2) tell - doxygen where are header files are, so include-file processing - can be done 3) make all macros that are used to define - variables/functions be expanded, so that doxygen will properly - document the resulting variable/function 4) make all macros that - are used to provide the contents of a variable (structure) be - expanded, so that doxygen will be able to document the resulting - fields 5) suppress compiler attributes (__attribute__(xxx)) from - being seen by doxygen, so it will properly match up function - definition and usage (for an example of th effect of this, look - at the doxygen docs for ast_log() from before and afte this - commit) ........ - -2008-11-14 17:03 +0000 [r156912] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /: Merged revisions 156911 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | - tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines - Ping is missing the standard double-newline after the event. - (closes issue #13903) Reported by: kebl0155 ........ - -2008-11-14 16:55 +0000 [r156818-156889] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/strings.h, apps/app_queue.c: This is the 1.6.0 - version of revision 156883 of trunk. This is different in that it - preserves the case-sensitiveness of processing queues from - configuration. closes issue #13703 - - * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 - (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov - 2008) | 10 lines If the prompt to reenter a voicemail password - timed out, it resulted in the password not being saved, even if - the input matched what you gave when first prompted to enter a - new password. This is because the return value of ast_readstring - was checked, but not checked properly. This bug was discovered by - Jared Smith during an Asterisk training course. Thanks for - reporting it! ........ ................ - -2008-11-13 19:26 +0000 [r156652-156653] Brandon Kruse <bkruse@digium.com> - - * main/manager.c: Update to Coding Guidelines - - * main/manager.c, /: Merged revisions 156017 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | - pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines - Patch by Ryan Brindley -- Make sure that manager refuses any - duplicate 'new category' requests in updateconfig (closes issue - #13539) ........ - -2008-11-12 19:56 +0000 [r156319] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 156299 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | - 26 lines Merged revisions 156297 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | - 18 lines It turns out that the 0x0XX00 codes being returned for - N, X, and Z are off by one, as per conversation with jsmith on - #asterisk-dev; he was teaching a class and disconcerted that this - published rule was not being followed, with patterns _NXX, - _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should - have been. This change, tested on these 3 patterns now picks the - proper one. However, this change may surprise users who set up - dialplans based on previous behavior, which has been there for - what, 2 and half years or so now. ........ ................ - -2008-11-12 18:57 +0000 [r156251] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 156243 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 - (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) - | 11 lines Revert revision 132506, since it occasionally caused - IAX2 HANGUP packets not to be sent, and instead, schedule a task - to destroy the iax2 pvt structure 10 seconds later. This allows - the IAX2 HANGUP packet to be queued, transmitted, and ACKed - before the pvt is destroyed. (closes issue #13645) Reported by: - dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by - Corydon76 (license 14) Tested by: vazir Reviewed: - http://reviewboard.digium.com/r/51/ ........ ................ - -2008-11-12 17:47 +0000 [r156170] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov - 2008) | 15 lines Merged revisions 156167 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov - 2008) | 7 lines When doing some tests, I was having a crash at - the end of every call if an attended transfer occurred during the - call. I traced the cause to the CDR on one of the channels being - NULL. murf suggested a check in the end bridge callback to be - sure the CDR is non-NULL before proceeding, so that's what I'm - adding. ........ ................ - -2008-11-11 21:28 +0000 [r156012] Russell Bryant <russell@digium.com> - - * apps/app_directory.c: Don't blow up if we get NULL when trying to - parse out the full name field (fixed for Jared in the training - room) - -2008-11-11 20:04 +0000 [r156007] Michiel van Baak <michiel@vanbaak.info> - - * /: remove prop that shouldn't be here - -2008-11-11 19:49 +0000 [r155815-156004] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | - tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines - Make documentation of update method match documentation and - update update2 method to match. Reported by: atis, via -dev - mailing list. Fixed by: me ........ - - * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | - tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line - I got tired of saying this in every single bugnote referring to - this file. ........ - -2008-11-09 01:34 +0000 [r155555] Sean Bright <sean.bright@gmail.com> - - * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, - apps/app_followme.c, apps/app_queue.c: Merged revisions 155554 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 - (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov - 2008) | 6 lines Use static functions here instead of nested ones. - This requires a small change to the ast_bridge_config struct as - well. To understand the reason for this change, see the following - post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html - ........ ................ - -2008-11-07 23:42 +0000 [r155361-155468] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | - mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 - lines Set the invite state to INV_CANCELLED in a place that makes - more sense. Where it was set before, it was impossible to - actually delay sending a CANCEL if we had not yet received a - provisional response to an INVITE. (closes issue #13626) Reported - by: atis Patches: 13626.patch uploaded by putnopvut (license 60) - Tested by: atis ........ - - * /, configs/voicemail.conf.sample: Merged revisions 155360 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, - 07 Nov 2008) | 8 lines Remove one more instance of the sample - configuration lying about what's possible. The tz cannot be set - in a context like this. It can only be set in the general section - or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing - this out ........ - -2008-11-06 22:50 +0000 [r155123] Kevin P. Fleming <kpfleming@digium.com> - - * /, res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: Merged - revisions 155121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | - kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 - lines don't blindly assume that Darwin and Cygwin need - GLOB_ABORTED defined; only define it if it is not already defined - ........ - -2008-11-06 19:47 +0000 [r155013] Mark Michelson <mmichelson@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 155012 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 - (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov - 2008) | 8 lines The documentation listed the ability to set - 'maxmsg' per context. The truth is that you can only set this in - the general section or per mailbox. Thus I am updating the sample - config file to be more accurate. Thanks to sasargen on IRC for - bringing up this issue. ........ ................ - -2008-11-03 22:30 +0000 [r154062-154081] Tilghman Lesher <tlesher@digium.com> - - * /: Recorded merge of revisions 154072 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) - | 12 lines Merged revisions 154066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) - | 5 lines Attempting to expunge a mailbox when the mailstream is - NULL will crash Asterisk. (Closes issue #13829) Reported by: - jaroth Patch by: me (modified jaroth's patch) ........ - ................ - - * main/rtp.c, /: Merged revisions 154060 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) - | 3 lines Remove the potential for a division by zero error. - (Closes issue #13810) ........ - -2008-11-03 00:53 +0000 [r153743-153746] Kevin P. Fleming <kpfleming@digium.com> - - * /: record revisions that were manually merged - - * apps/app_stack.c, include/asterisk/agi.h, configure, - include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, - configure.ac, include/asterisk/compiler.h: Merge revision 153709 - from trunk - ------------------------------------------------------------------------ - r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov - 2008) | 3 lines instead of trying to forcibly load res_agi when - app_stack is loaded (even if the administrator didn't want it - loaded), use GCC weak symbols to determine whether it was loaded - already or not; if it was loaded, then use it. - ------------------------------------------------------------------------ - - * channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c, - channels/chan_h323.c, main/file.c, apps/app_sms.c, - pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c, - apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, - main/asterisk.c, apps/app_voicemail.c, utils/muted.c, - apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c, - res/res_musiconhold.c, formats/format_wav_gsm.c, - res/res_jabber.c, channels/chan_iax2.c, utils/frame.c, - utils/stereorize.c, main/channel.c, channels/chan_dahdi.c, - main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c, - main/ast_expr2f.c, res/res_agi.c, main/logger.c, main/http.c, - formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c, - channels/chan_sip.c, formats/format_wav.c, apps/app_festival.c, - main/db1-ast/hash/hash_page.c, res/ael/ael.y, res/res_crypto.c, - agi/eagi-test.c, utils/astman.c, pbx/pbx_lua.c, - formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c: - port gcc 4.3.x warning fixes from trunk to this branch - -2008-10-31 21:49 +0000 [r153265] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, - apps/app_followme.c, apps/app_queue.c: Merged revisions 153181 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 - Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' - exten into the bridging code, so variables that were set after - ast_bridge_call was called would not show up in the 'h' exten. - Added a callback function to handle setting variables, etc. from - w/in the bridging code. Calls back into a nested function within - the function calling ast_bridge_call (closes issue #13793) - Reported by: greenfieldtech ........ - -2008-10-30 21:00 +0000 [r152994] Sean Bright <sean.bright@gmail.com> - - * /, bootstrap.sh: Merged revisions 152993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct - 2008) | 10 lines Merged revisions 152992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct - 2008) | 2 lines The -I argument to aclocal needs a space before - the include directory name. ........ ................ - -2008-10-30 16:54 +0000 [r152813] Kevin P. Fleming <kpfleming@digium.com> - - * main/cdr.c, /: Merged revisions 152812 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct - 2008) | 9 lines Merged revisions 152811 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct - 2008) | 3 lines instead of comparing the string pointer to 0, - let's compare the value that was actually parsed out of the - string (found by sparse) ........ ................ - -2008-10-30 04:28 +0000 [r152772] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 152765 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29 - Oct 2008) | 5 lines Set up an example stdexten that preserves the - original context and extension in the CDR. (Related to issue - #13799) Reported by: davidw ........ - -2008-10-29 20:54 +0000 [r152647] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_directory.c: Merged revisions 152646 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct - 2008) | 9 lines If there was no named defined in a voicemail.conf - mailbox entry, then app_directory would crash when attempting to - read that entry from the file. We now check for the NULL or empty - string properly so that there will be no crash. (closes issue - #13804) Reported by: bluecrow76 ........ - -2008-10-29 20:13 +0000 [r152644] Terry Wilson <twilson@digium.com> - - * apps/app_queue.c: Small modification to putnopvut's patch to fix - this issue. Thanks for all the help, putnopvut! (closes issue - #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch - uploaded by otherwiseguy (license 396) Tested by: otherwiseguy - -2008-10-28 21:39 +0000 [r152443] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_mgcp.c, /: Merged revisions 152442 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008) - | 7 lines Only re-add the io port if it was closed, otherwise - reload causes a memory leak. (closes issue #13785) Reported by: - eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) - ........ - -2008-10-27 16:33 +0000 [r152157] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c, /: Merged revisions 152134 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 | - tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines - Oops, only delete the ARG variables once upon release. The - following section would have removed them again (removing - variables from 2 stack frames, instead of just one). ........ - -2008-10-26 20:26 +0000 [r152062] Sean Bright <sean.bright@gmail.com> - - * /, funcs/func_strings.c: Merged revisions 152060 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r152060 | seanbright | 2008-10-26 16:25:08 -0400 - (Sun, 26 Oct 2008) | 15 lines Merged revisions 152059 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct - 2008) | 7 lines Since passing \0 as the second argument to strchr - is valid (and will match the trailing \0 of a string) we need to - check that first, otherwise we end up with incorrect results. Fix - suggested by reporter. (closes issue #13787) Reported by: - meitinger ........ ................ - -2008-10-23 16:12 +0000 [r151765] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix some memory leaks. These issues are - 1.6.0 specific. - Freeing the peer got accidentally removed from - the peer's destructor. It is still needed for astobj, but not for - astobj2. - Fix some places that called find_user or find_peer, - but did not release the reference that was returned. (closes - issue #13331) Reported by: sergee Patches: - chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138) - Tested by: sergee - -2008-10-20 05:03 +0000 [r151244] Kevin P. Fleming <kpfleming@digium.com> - - * autoconf (added), autoconf/ast_check_pwlib.m4, - autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, - autoconf/ast_gcc_attribute.m4, bootstrap.sh, - autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, - autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, - autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, - autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, - /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, - configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: - Merged revisions 151242-151243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct - 2008) | 9 lines Merged revisions 151240 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct - 2008) | 3 lines break up acinclude.m4 into individual files, - which will make it easier to maintain, easier to add new macros - (less patching) and will ease maintenance of these macros across - Asterisk branches ........ ................ r151243 | kpfleming | - 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines Merged - revisions 151241 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct - 2008) | 2 lines rename this macro to properly reflect what it - does ........ ................ - -2008-10-18 02:35 +0000 [r150854] BJ Weschke <bweschke@btwtech.com> - - * main/manager.c, /: Merged revisions 150817 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r150817 | - bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines - Using the GetVar handler in AMI is potentially dangerous - (insta-crash [tm]) when you use a dialplan function that requires - a channel and then you don't provide one or provide an invalid - one in the Channel: parameter. We'll handle this situation - exactly the same way it was handled in pbx.c back on r61766. - We'll create a bogus channel for the function call and destroy it - when we're done. If we have trouble allocating the bogus channel - then we're not going to try executing the function call at all - and run the risk of crashing. (closes issue #13715) reported by: - makoto patch by: bweschke ........ - -2008-10-17 00:19 +0000 [r150308-150313] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Instead of merging commit 150307 to 1.6.0, I - had meant to block it in 1.6.1...time to go home :) - - * /, channels/chan_sip.c: Merged revisions 150307 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 | - mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 - lines After a long discussion on #asterisk-bugs, it seems kind of - odd that a channel would be named after the port on which it came - in on. For endpoints that always include ":5060" as part of the - From: header, it will mean that you have a ton of channels with - names like "SIP/5060-3ea38a8b." I am boldly moving forward with - this change in trunk, but I'm not touching other branches with - this one since this definitely would qualify as a behavior - change. If there is a problem with this commit, and I haven't - seen the obvious reason why you'd want to name the channel after - the port from which the call originated, then please feel free to - revert this ........ - -2008-10-16 16:10 +0000 [r150126] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c, /: Merged revisions 150125 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r150125 | rmudgett | 2008-10-16 11:04:45 -0500 - (Thu, 16 Oct 2008) | 9 lines Merged revisions 150124 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 - Oct 2008) | 1 line Fix memory leak found by customer ........ - ................ - -2008-10-15 20:18 +0000 [r149757] BJ Weschke <bweschke@btwtech.com> - - * configs/agents.conf.sample, /: Merged revisions 149756 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400 - (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) - | 4 lines An update to the documentation/example of - agents.conf.sample with the correct parameter for this feature as - defined in chan_agent.c (closes issue #13709) ........ - ................ - -2008-10-15 11:29 +0000 [r149495] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 149487 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct - 2008) | 9 lines Merged revisions 149452 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct - 2008) | 3 lines fix some problems when parsing SIP messages that - have the maximum number of headers or body lines that we support - ........ ................ - -2008-10-14 17:39 +0000 [r148914] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: Merged revisions 148913 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r148913 | mmichelson | 2008-10-14 12:38:06 -0500 - (Tue, 14 Oct 2008) | 17 lines Merged revisions 148912 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct - 2008) | 9 lines Deadlock prevention in chan_local. (closes issue - #13676) Reported by: tacvbo Patches: 13676.patch uploaded by - putnopvut (license 60) Tested by: tacvbo ........ - ................ - -2008-10-14 10:34 +0000 [r148613-148739] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 148738 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct - 2008) | 9 lines Merged revisions 148736 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct - 2008) | 3 lines on Ubuntu (at least), recent versions of ld in - binutils delete all debugging symbols when -x is supplied; since - the reasons why -x is being passed are lost in the mists of time, - remove it so debugging will work properly ........ - ................ - - * /, main/translate.c: Merged revisions 148612 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct - 2008) | 9 lines Merged revisions 148611 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct - 2008) | 3 lines it would be nice if this message printing code - had actually been tested before it was committed... ........ - ................ - -2008-10-10 21:18 +0000 [r148374] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 148373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 | - mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 - lines Make sure that the inUse and inRinging fields for a sip - peer cannot go below zero. This is a regression from 1.4 and so - it will be applied to 1.6.0 as well. (closes issue #13668) - Reported by: mjc ........ - -2008-10-10 01:25 +0000 [r148201-148204] Sean Bright <sean.bright@gmail.com> - - * res/res_config_sqlite.c, apps/app_voicemail.c, - include/asterisk.h, /, main/tdd.c, main/cryptostub.c: Merged - revisions 148200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 | - seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 - lines Don't include logger.h in asterisk.h by default as it is - causing problems building app_voicemail. Instead, include it - where it is needed. This turned out to be a relatively minor - issue because other headers include logger.h as well. Need to - test -addons before merging this back to 1.6.0. (closes issue - #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff - uploaded by seanbright (license 71) Tested by: mmichelson - ........ - - * apps/app_rpt.c: Somehow we got conflict markers checked in! Might - need a 1.6.0.1 sooner than we'd like. - -2008-10-09 23:31 +0000 [r148147] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 148144 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct - 2008) | 10 lines Read the callerid in the correct order and make - sure to read the Urgent flag value from the IMAP headers. (closes - issue #13652) Reported by: jaroth Patches: imapheaders.patch - uploaded by jaroth (license 50) ........ - -2008-10-09 23:26 +0000 [r148124] Tilghman Lesher <tlesher@digium.com> - - * /, configs/res_ldap.conf.sample: Merged revisions 148120 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09 - Oct 2008) | 6 lines Fix example schema (closes issue #12860) - Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn - (license 503) ........ - -2008-10-09 17:51 +0000 [r147900] Michiel van Baak <michiel@vanbaak.info> - - * include/asterisk/endian.h, /: Merged revisions 147899 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09 - Oct 2008) | 5 lines only include this for OpenBSD. At least - FreeBSD is borked when including it (closes issue #13649) - Reported by: ys ........ - -2008-10-09 17:47 +0000 [r147897] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 147896 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09 - Oct 2008) | 4 lines Remove "second form" of extensions, as it no - longer applies. Also, cleanup the grammar, formatting, and - introduce several clarifications to the text. (Closes issue - #13654) ........ - -2008-10-09 14:56 +0000 [r147809] Steve Murphy <murf@digium.com> - - * main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c, - main/cli.c, configure, channels/console_gui.c, utils/extconf.c, - main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, /, - include/asterisk/autoconfig.h.in, main/translate.c, - channels/vcodecs.c, configure.ac, channels/console_video.c, - channels/chan_iax2.c: Merged revisions 147807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | - murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines - (closes issue #13557) Reported by: nickpeirson Patches: - pbx.c.patch uploaded by nickpeirson (license 579) - replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) - Tested by: nickpeirson, murf 1. replaced all refs to bzero and - bcopy to memset and memmove instead. 2. added a note to the - CODING-GUIDELINES 3. add two macros to asterisk.h to prevent - bzero, bcopy from creeping back into the source 4. removed bzero - from configure, configure.ac, autoconfig.h.in ........ - -2008-10-08 12:16 +0000 [r147458] Russell Bryant <russell@digium.com> - - * configs/chan_dahdi.conf.sample: Remove the sample configuration - for configuration sections in chan_dahdi.conf. This code was not - merged into 1.6.0. Reported by: angler (closes AST-119) - -2008-10-08 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0.1 released. - - * configs/chan_dahdi.conf.sample: Remove mention of configuration - sections for defining channels in chan_dahdi.conf. This code - is in 1.6.1, and was not merged into 1.6.0. - -2008-10-01 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0 released. - -2008-09-09 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-rc6 released. - -2008-09-09 15:44 +0000 [r142065] Russell Bryant <russell@digium.com> - - * /, main/features.c: Merged revisions 142064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) - | 13 lines Merged revisions 142063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) - | 5 lines Ensure that the stored CDR reference is still valid - after the bridge before poking at it. Also, keep the channel - locked while messing with this CDR. (fixes crashes reported in - issue #13409) ........ ................ - -2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson <mmichelson@digium.com> - - * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | - mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 - lines Fix a memory leak in chan_oss (closes issue #13311) - Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel - (license 64) ........ - -2008-09-09 01:49 +0000 [r141950] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 141949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | - russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines - Modify ast_answer() to not hold the channel lock while calling - ast_safe_sleep() or when calling ast_waitfor(). These are - inappropriate times to hold the channel lock. This is what has - caused "could not get the channel lock" messages from chan_sip - and has likely caused a negative impact on performance results of - SIP in Asterisk 1.6. Thanks to file for pointing out this section - of code. (closes issue #13287) (closes issue #13115) ........ - -2008-09-08 21:07 +0000 [r141808] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 141807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) - | 15 lines Merged revisions 141806 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) - | 7 lines When doing an async goto, detect if the channel is - already in the middle of a masquerade. This can happen when - chan_local is trying to optimize itself out. If this happens, - fail the async goto instead of bursting into flames. (closes - issue #13435) Reported by: geoff2010 ........ ................ - -2008-09-08 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-rc5 released. - -2008-09-08 20:19 +0000 [r141746] Jason Parker <jparker@digium.com> - - * Makefile, /, redhat (removed): Merged revisions 141745 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 - (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | - 8 lines Remove RPM package targets from Makefile (and all - associated parts). This has never worked in 1.4, and we decided - that it makes no sense to be done here. There are many distros - out there that already have "proper" spec files that can be - (re)used. Closes issue #13113 Closes issue #10950 Closes issue - #10952 ........ ................ - -2008-09-08 17:14 +0000 [r141683] Sean Bright <sean.bright@gmail.com> - - * /, build_tools/make_buildopts_h: Merged revisions 141682 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, - 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on - various platforms doesn't choke on the special characters (like - ^). (closes issue #13417) Reported by: dougm Patches: - 13417.make_buildopts_h.patch uploaded by seanbright (license 71) - Tested by: dougm ........ - -2008-09-06 20:21 +0000 [r141567] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 - lines Merged revisions 141565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 - line This fix comes from Joshua Colp The Brilliant, who, given - the trace, came up with a solution. This will most likely will - close 13235 and 13409. I'll wait till Monday to verify, and then - close these bugs. ........ ................ - -2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 141504 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) - | 12 lines Merged revisions 141503 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) - | 4 lines Reverting behavior change (AGI should not exit non-zero - on SUCCESS) (closes issue #13434) Reported by: francesco_r - ........ ................ - -2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 - (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep - 2008) | 7 lines Agent's should not try to call a channel's - indicate callback if the channel has been hung up. It will likely - crash otherwise ABE-1159 ........ ................ - -2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy <murf@digium.com> - - * main/channel.c, /: Merged revisions 141157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 - lines Merged revisions 141156 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 - line A small change to prevent double-posting of CDR's; thanks to - Daniel Ferrer for bringing it to our attention ........ - ................ - - * pbx/ael/ael-test/ref.ael-vtest25 (added), /, - pbx/ael/ael-test/ael-vtest25/extensions.ael, - pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, - pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged - revisions 141115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | - 78 lines Merged revisions 141094 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | - 70 lines (closes issue #13357) Reported by: pj Tested by: murf - (closes issue #13416) Reported by: yarns Tested by: murf If you - find this message overly verbose, relax, it's probably not meant - for you. This message is meant for probably only two people in - the whole world: me, or the poor schnook that has to maintain - this code because I'm either dead or unavailable at the moment. - This fix solves two reports, both having to do with embedding a - function call in a ${} construct. It was tricky because the - funccall syntax has parenthesis () in it. And up till now, the - 'word' token in the flex stuff didn't allow that, because it - would tend to steal the LP and RP tokens. To be truthful, the - "word" token was the trickiest, most unstable thing in the whole - lexer. I was lucky it made this long without complaints. I had to - choose every character in the pattern with extreme care, and I - knew that someday I'd have to revisit it. Well, the day has come. - So, my brilliant idea (and I'm being modest), was to use the - surrounding ${} construct to make a state machine and capture - everything in it, no matter what it contains. But, I have to now - treat the word token like I did with comments, in that I turn the - whole thing into a state-machine sort of spec, with new contexts - "curlystate", "wordstate", and "brackstate". Wait a minute, - "brackstate"? Yes, well, it didn't take very many regression - tests to point out if I do this for ${} constructs, I also have - to do it with the $[] constructs, too. I had to create a separate - pcbstack2 and pcbstack3 because these constructs can occur inside - macro argument lists, and when we have two state machines - operating on the same structures we'd get problems otherwise. I - guess I could have stopped at pcbstack2 and had the brackstate - stuff share it, but it doesn't hurt to be safe. So, the pcbpush - and pcbpop routines also now have versions for "2" and "3". I had - to add the {KEYWORD} construct to the initial pattern for "word", - because previously word would match stuff like "default7", - because it was a longer match than the keyword "default". But, - not any more, because the word pattern only matches only one or - two characters now, and it will always lose. So, I made it the - winner again by making an optional match on any of the keywords - before it's normal pattern. I added another regression test to - make sure we don't lose this in future edits, and had to fix just - one regression, where it no longer reports a 'cascaded' error, - which I guess is a plus. I've given some thought as to whether to - apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I - decided to put it in 1.4 because one of the bug reports was - against 1.4; and it is unexpected that AEL cannot handle this - situation. It actually reduced the amount of useless "cascade" - error messages that appeared in the regressions (by one line, - ehhem). There is a possible side-effect in that it does now do - more careful checking of what's in those ${} constructs, as far - as matching parens, and brackets are concerned. Some users may - find a an insidious problem and correct it this way. This should - be exceedingly rare, I hope. ........ ................ - -2008-09-04 18:35 +0000 [r141086] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c, res/res_agi.c: Merged revisions 141039 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 - (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) - | 7 lines (closes issue #11979) Fixes multiple parking problems: - Crash when executing a park on an extension dialed by AGI due to - not returning the proper return code. Crash when using a builtin - feature that was a subset of a enabled dynamic feature. Crash due - to always hanging up the peer despite the fact that the peer was - supposed to be parked. ........ ................ - -2008-09-03 20:18 +0000 [r140976] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | - mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 - lines Fix some locking order issues in app_queue. This was - brought up by atis on IRC a while ago. ........ - -2008-09-03 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-rc4 released. - -2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 140749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | - 11 lines Merged revisions 140747 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 - line I am turning the warnings generated in ast_cdr_free and - post_cdr into verbose level 2 messages. Really, they matter - little to end users. You either get the CDR's you wanted, or you - don't, and it is a bug. For trunk, I am going one step further. - These messages were pretty worthless even for debug, so I'm - completely removing them. ........ ................ - - * main/channel.c, /: Merged revisions 140692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | - 13 lines Merged revisions 140690 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 - line After reconsidering, with respect to 13409, ast_cdr_detach - should be OK, better in fact, than ast_cdr_free, which generates - lots of useless warnings that will undoubtably generate - complaints. Hmmm. It doesn't hush the useless warnings, but it - does allow control of posting via the detach and post routines, - for those possible situations, where you'd want to post - single-channel cdrs. ........ ................ - - * main/channel.c, main/pbx.c, /: Merged revisions 140691 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, - 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | - 14 lines (closes issue #13409) Reported by: tomaso Patches: - asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license - 564) I basically spent the day, verifying that this patch solves - the problem, and doesn't hurt in non-problem cases. Why valgrind - did not plainly reveal this leak absolutely mystifies and stuns - me. Many, many thanks to tomaso for finding and providing the - fix. ........ ................ - -2008-09-03 13:27 +0000 [r140818] Russell Bryant <russell@digium.com> - - * main/poll.c, /: Merged revisions 140817 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) - | 12 lines Merged revisions 140816 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) - | 4 lines Don't freak out if the poll emulation receives NULL for - the pollfds array (closes issue #13307) Reported by: jcovert - ........ ................ - -2008-09-02 18:17 +0000 [r140607] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 - (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep - 2008) | 8 lines Make sure to use the correct length of the - mohinterpret and mohsuggest buffers when copying configuration - values. (closes issue #13336) Reported by: - decryptus_proformatique Patches: - chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded - by decryptus (license 555) ........ ................ - -2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant <russell@digium.com> - - * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions - 140566 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | - russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines - Update instructions for getting libresample ........ - -2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Revert commit 140302. Should not be merging - changes like that into a release-candidate branch - - * channels/chan_sip.c: Merged revisions 140301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug - 2008) | 19 lines Merged revisions 140299 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug - 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when - in pedantic mode. The problem was that the wrong tags would be - compared depending on the direction of the call. (closes issue - #13353) Reported by: flefoll Patches: - chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll - (license 244) ........ ................ - -2008-08-26 18:12 +0000 [r140170] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 140169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | - russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines - Fix building menuselect-tree with PRINT_DIR set. We _must_ use - the --quiet flag here, or else some arbitrary text will end up in - the resulting menuselect-tree file and things will explode. - ........ - -2008-08-25 21:33 +0000 [r139918] Sean Bright <sean.bright@gmail.com> - - * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged - revisions 139915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug - 2008) | 17 lines Merged revisions 139909 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug - 2008) | 9 lines Some versions of awk (nawk, for example) don't - like empty regular expressions so be slightly more verbose. - (closes issue #13374) Reported by: dougm Patches: 13374.diff - uploaded by seanbright (license 71) Tested by: dougm ........ - ................ - -2008-08-25 21:05 +0000 [r139872] Terry Wilson <twilson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) - | 10 lines Merged revisions 139869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) - | 2 lines Make SIPADDHEADER() propagate indefinitely ........ - ................ - -2008-08-25 16:00 +0000 [r139774] Steve Murphy <murf@digium.com> - - * main/pbx.c, /, main/features.c: Merged revisions 139770 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, - 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 - lines This patch reverts the changes made via 139347, and 139635, - as users are seeing adverse difference. I will un-close 13251. - Back to the drawing board/ concept/ beginning/ whatever! ........ - ................ - -2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | - tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines - Memory leak ........ - -2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 139662 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | - 14 lines Merged revisions 139635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 - lines I found some problems with the code I committed earlier, - when I merged them into trunk, so I'm coming back to clean up. - And, in the process, I found an error in the code I added to - trunk and 1.6.x, that I'll fix using this patch also. ........ - ................ - - * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions - 139627 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | - 59 lines Merged revisions 139347 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | - 47 lines (closes issue #13251) Reported by: sergee Tested by: - murf THis is a bold move for a static release fix, but I wouldn't - have made it if I didn't feel confident (at least a *bit* - confident) that it wouldn't mess everyone up. The reasoning goes - something like this: 1. We simply cannot do anything with CDR's - at the current point (in pbx.c, after the __ast_pbx_run loop). - It's way too late to have any affect on the CDRs. The CDR is - already posted and gone, and the remnants have been cleared. 2. I - was very much afraid that moving the running of the 'h' extension - down into the bridge code (where it would be now practical to do - it), would result in a lot more calls to the 'h' exten, so I - implemented it as another exten under another name, but found, to - my pleasant surprise, that there was a 1:1 correspondence to the - running of the 'h' exten in the pbx_run loop, and the new spot at - the end of the bridge. So, I ifdef'd out the current 'h' loop, - and moved it into the bridge code. The only difference I can see - is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this - is still an important decision point, I can replicate it if there - are complaints. To be perfectly honest, the KEEPALIVE situation - is not totally clear to me, and how it relates to a post-bridge - situation is less clear. I suspect the users will point out - everything in total clarity if this steps on anyone's toes! 3. I - temporarily swap the bridge_cdr into the channel before running - the 'h' exten, which makes it possible for users to edit the cdr - before it goes out the door. And, of course, with the - endbeforehexten config var set, the users can also get at the - billsec/duration vals. After the h exten finishes, the cdr is - swapped back and processing continues as normal. Please, all who - deal with CDR's, please test this version of Asterisk, and file - bug reports as appropriate! ........ I also made a little fix to - the app_dial's 'e' option, that is related to my updates. - ................ - -2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/threadstorage.h, /: Merged revisions 139554 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 - (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug - 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is - selected (closes issue #13298) Reported by: snuffy Patches: - bug13298_20080822.diff uploaded by snuffy (license 35) ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 - (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug - 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 - (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug - 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from - incorrect locking order between iax2_pvt and ast_channel - structures. AST-13 ........ ................ - -2008-08-21 23:46 +0000 [r139400] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 - (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) - | 3 lines Fixes loop that could possibly never exit in the event - of a channel never being able to be opened or specify after a - restart. (closes issue #11017) ........ ................ - -2008-08-21 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) - | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! - (closes issue #13310) Reported by: eliel Patches: - chan_gtalk.c.patch uploaded by eliel (license 64) ........ - -2008-08-20 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.6.0-rc3 released. - -2008-08-20 22:17 +0000 [r139216] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) - | 19 lines Merged revisions 139213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) - | 11 lines Fix a crash in the ChanSpy application. The issue here - is that if you call ChanSpy and specify a spy group, and sit in - the application long enough looping through the channel list, you - will eventually run out of stack space and the application with - exit with a seg fault. The backtrace was always inside of a - harmless snprintf() call, so it was tricky to track down. - However, it turned out that the call to snprintf() was just the - biggest stack consumer in this code path, so it would always be - the first one to hit the boundary. (closes issue #13338) Reported - by: ruddy ........ ................ - -2008-08-20 20:12 +0000 [r139155] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: Fix bug where the samples were not accurate - when in G723 mode, which would cause the timestamp field of the - RTP header to be invalid. - -2008-08-20 17:30 +0000 [r139104] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 139083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | - 20 lines Merged revisions 139074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | - 12 lines (closes issue #13263) Reported by: brainy Tested by: - murf The specialized reset routine is tromping on the flags field - of the CDR. I made a change to not reset the DISABLED bit. This - should get rid of this problem. ........ ................ - -2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug - 2008) | 14 lines Merged revisions 139015 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug - 2008) | 6 lines sip_read should properly handle a NULL return - from sip_rtp_read. (closes issue #13257) Reported by: travishein - ........ ................ - - * apps/app_chanspy.c: Manually add revision 138887 from trunk to - the 1.6.0 branch. I had misunderstood the policy for when to - merge to 1.6.0 since it moved to rc status. - -2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy <murf@digium.com> - - * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, - res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug - 2008) | 1 line Oops. put a decl in a generated file. My bad, but - fixed now. ........ - - * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, - res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | - murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines - These changes are in regards to bug 13249, where users are being - surprised by the changes made to the Set app in trunk/1.6.x, as - they come from the 1.4 world. They are only bitten if they write - their AEL dialplan in the 1.4 world, and then carry it over to a - trunk/1.6.x installation where a "make samples" was executed, or - where they hand-edited the asterisk.conf file and added the - [compat] category with app_set = 1.6 (or higher). (this commit - does not totally solve 13249, at least not yet) The change - involves issueing a single warning while the AEL file is loading, - if: 1. app_set is present in the config file, and set to 1.6 or - higher. 2. there are double quotes in an assignment statement (eg - x = "hi there";) 3. the warning was not already issued. The - standalone app, aelparse, does not (yet) issue this warning. I'd - have to have it read in the asterisk.conf file, and that's a bit - of hassle. I'll add it if users request it, tho. ........ - -2008-08-19 00:15 +0000 [r138776-138781] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_sip.c: Merged revisions 138778-138780 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, - 18 Aug 2008) | 1 line While we're at it, make this machine - parseable too. ........ r138779 | seanbright | 2008-08-18 - 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we - don't need anymore. ........ r138780 | seanbright | 2008-08-18 - 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, - too (woops) ........ - - * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | - seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 - lines Change event header to RegistrationTime to be more - consistent (and avoid breaking existing frameworks). Pointed out - by Laureano on #asterisk-dev. ........ - -2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug - 2008) | 18 lines Merged revisions 138685 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug - 2008) | 10 lines Change the inequalities used in app_queue with - regards to timeouts from being strict to non-strict for more - accuracy. (closes issue #13239) Reported by: atis Patches: - app_queue_timeouts_v2.patch uploaded by atis (license 242) - ........ ................ - -2008-08-18 15:54 +0000 [r138632] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 138631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | - qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line - Remove option that isn't valid here. ........ - -2008-08-18 02:14 +0000 [r138519] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) - | 1 line add missing define for SS7 in dahdi_restart ........ - -2008-08-17 14:14 +0000 [r138443-138483] Sean Bright <sean.bright@gmail.com> - - * /, main/features.c: Merged revisions 138482 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | - seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 - lines Move Uniqueid to the end of the event for those that rely - on the position of the name/value pairs, pointed out by - snuffy-home on #asterisk-commits. For those of you who rely on - the position of name/value pairs in manager events... stop... - that is why associative arrays were invented. ........ - - * /, main/features.c: Merged revisions 138479 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | - seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 - lines Add Uniqueid header to ParkedCall manager event. (closes - issue #13323) Reported by: srt Patches: - 13323_unique_id_for_parkedcalls_event.diff uploaded by srt - (license 378) ........ - - * main/rtp.c, /: Merged revisions 138476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | - seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 - lines Add missing colons to RTCPReceived and RTCPSent manager - events. (closes issue #13319) Reported by: srt Patches: - 13319_rtcp_manager_event_headers.diff uploaded by srt (license - 378) ........ - - * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug - 2008) | 7 lines Fix the output of the JitterBufStats manager - event. (closes issue #13324) Reported by: srt Patches: - 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt - (license 378) ........ - - * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, - 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, - cdr_tds has *never* read the port configuration option from - cdr_tds.conf. So go ahead and remove it from the sample config. - ........ - -2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) - | 2 lines Fix compilation warnings (found with dev-mode) ........ - -2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 - (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 - Aug 2008) | 1 line fixes use count to properly decrement if an - active dahdi channel is destroyed allowing module to be unloaded - ........ ................ - - * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 - (Fri, 15 Aug 2008) | 20 lines Merged revisions - 138119,138151,138238 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) - | 4 lines Fixes the dahdi restart functionality. Dahdi restart - allows one to restart all DAHDI channels, even if they are - currently in use. This is different from unloading and then - loading the module since unloading requires the use count to be - zero. Reloading the module is different in that the signalling is - not changed from what it was originally configured. Also, this - fixes not closing all the file descriptors for D-channels upon - module unload (which would prevent loading the module - afterwards). (closes issue #11017) ........ r138151 | jpeeler | - 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared - static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ - r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) - | 1 line initialize condition variable ss_thread_complete using - ast_cond_init ........ ................ - -2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 138260 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) - | 16 lines Merged revisions 138258 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) - | 8 lines More fixes for realtime peers. (closes issue #12921) - Reported by: Nuitari Patches: 20080804__bug12921.diff.txt - uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt - uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ - ................ - - * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions - 138206 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | - tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines - Remove deprecated syntax from sample config file (closes issue - #13314) Reported by: kue ........ - -2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to - dfd to match 1.4 (left over from DAHDI transition) - -2008-08-15 15:12 +0000 [r138029] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: Merged revisions 138028 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) - | 17 lines Merged revisions 138027 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) - | 9 lines Ensure that when a hangup occurs in autoservice, that a - hangup frame gets properly deferred to be read from the channel - owner when it gets taken out of autoservice. (closes issue - #12874) Reported by: dimas Patches: v1-12874.patch uploaded by - dimas (license 88) ........ ................ - -2008-08-15 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 - (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) - | 8 lines Additional check for more string specifiers than - arguments. (closes issue #13299) Reported by: adomjan Patches: - 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) - func_strings.c-sprintf.patch uploaded by adomjan (license 487) - Tested by: adomjan ........ ................ - -2008-08-14 22:43 +0000 [r137988] Russell Bryant <russell@digium.com> - - * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | - russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines - Fix a bashism that causes an error when trying to build the pdf - on ubuntu ........ - -2008-08-14 18:48 +0000 [r137934] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug - 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes - issue #13304) Reported by: eliel Patches: sqlite.patch uploaded - by eliel (license 64) (Slightly modified by me) ........ - -2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 - (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) - | 9 lines When creating the secondary subchannel name, it is - necessary to compare to the existing channel name without the - "Zap/" or "DAHDI/" prefix, since our test string is also without - that prefix. (closes issue #13027) Reported by: dferrer Patches: - chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) - (Slightly modified by me, to compensate for both names) ........ - ................ - -2008-08-14 Jason Parker <jparker@digium.com> - - * Asterisk 1.6.0-rc2 released. - -2008-08-14 15:37 +0000 [r137814] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | - qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines - Make sure we set the socket port, so we don't try to use <ip - address>:0. (closes issue #13255) Reported by: falves11 Patches: - 13255-socketport.diff uploaded by qwell (license 4) Tested by: - falves11 ........ - -2008-08-14 15:20 +0000 [r137783] Russell Bryant <russell@digium.com> - - * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r137732 | russell | 2008-08-14 09:15:50 -0500 - (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) - | 4 lines Comments in this config file were aligned only if your - tab size was set to 8. So, convert tabs to spaces so that things - should be aligned regardless of what tab size you use in your - editor. ........ ................ - -2008-08-14 15:05 +0000 [r137781] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | - seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 - lines If we detect that we are no longer connected, try to - reconnect a few times before giving up. This relies on the - timeout settings in the freetds.conf file and, unfortunately, on - a recent version of FreeTDS (0.82 or newer). I either need to - change the current execs to be non-blocking (which I do not want - to do) or we have to force people to run with the latest and - greatest of FreeTDS. I'm on the fence... ........ - -2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming <kpfleming@digium.com> - - * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug - 2008) | 9 lines Merged revisions 137679 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug - 2008) | 1 line forgot one module name that changed ........ - ................ - -2008-08-13 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.6.0-rc1 released. - -2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug - 2008) | 1 line make this script actually work ........ - - * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions - 137627 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug - 2008) | 9 lines Merged revisions 137530 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug - 2008) | 1 line add document describing what users will need to be - aware of when upgrading to this version and using DAHDI ........ - ................ - -2008-08-13 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | - qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines - Correctly end locally ended calls. (closes issue #12170) Reported - by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff - uploaded by bbryant (license 36) Tested by: bbryant, pabelanger - ........ - - * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | - qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines - Add FAXMODE variable with what fax transport was used. (closes - issue #13252) Patches: v1-13252.patch uploaded by dimas (license - 88) ........ - -2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com> - - * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 - (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, - 13 Aug 2008) | 1 line Update docs to reflect the change to - cdr_tds ........ ................ - - * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | - seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 - line Use the ast_vasprintf macro instead of vasprintf directly. - ........ - -2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com> - - * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) - | 2 lines Grammar hax from Qwell ........ - - * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) - | 3 lines Note that developer documentation belongs in doxygen, - and not integrated with the user manual stuff in doc/tex/. - ........ - -2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 137239 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | - russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines - Make PRINT_DIR work as advertised. ........ - -2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, - 11 Aug 2008) | 7 lines Log the userfield CDR variable like the - other CDR backends, assuming the column is actually there. If - it's not, we still log everything else as before. (closes issue - #13281) Reported by: falves11 ........ - -2008-08-11 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) - | 13 lines Merged revisions 137138 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) - | 5 lines Deallocate database connection handle on disconnect, as - we allocate another one on connect. (closes issue #13271) - Reported by: dveiga ........ ................ - -2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged - revisions 136947 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) - | 18 lines Merged revisions 136946 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 - (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) - | 2 lines Regression fixes for Solaris ........ ................ - ................ - -2008-08-09 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 136859 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | - tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines - Update documentation as to the behavior of AGI in 1.6.0 and - higher. Also, add an OOB message that answers the question of, if - AGI no longer shuts down the connection on hangup, how will - FastAGI know when to stop processing the call? ........ - -2008-08-08 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug - 2008) | 3 lines Fix compilation for ODBC voicemail ........ - -2008-08-08 06:45 +0000 [r136778] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /, - pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, - utils/ael_main.c: Merged revisions 136746 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | - 40 lines Merged revisions 136726 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | - 32 lines (closes issue #13236) Reported by: korihor Wow, this one - was a challenge! I regrouped and ran a new strategy for setting - the ~~MACRO~~ value; I set it once per extension, up near the - top. It is only set if there is a switch in the extension. So, I - had to put in a chunk of code to detect a switch in the pval - tree. I moved the code to insert the set of ~~exten~~ up to the - beginning of the gen_prios routine, instead of down in the switch - code. I learned that I have to push the detection of the switches - down into the code, so everywhere I create a new exten in - gen_prios, I make sure to pass onto it the values of the - mother_exten first, and the exten next. I had to add a couple - fields to the exten struct to accomplish this, in the - ael_structs.h file. The checked field makes it so we don't repeat - the switch search if it's been done. I also updated the - regressions. ........ ................ - -2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com> - - * /: Merged revisions 136751 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | - tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines - Removing bad properties ........ - -2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a - bunch of functions over one level during a merge. - - * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug - 2008) | 3 lines Remove one last batch of debug messages ........ - - * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug - 2008) | 18 lines Merging the imap_consistency_trunk branch to - trunk. For an explanation of what "imap_consistency" is, please - see svn revision 134223 to the 1.4 branch. Coincidentally, this - also fixes a recent bug report regarding the inability to save - messages to the new folder when using IMAP storage since they - will would be flagged as "seen" and not be recognized as new - messages. (closes issue #13234) Reported by: jaroth ........ - -2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: Removing code that was commented out. - - * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder - interface in the DAHDI. (Issue: DAHDI-42) - -2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com> - - * /, main/features.c: Merged revisions 136660 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | - mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 - lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears - once for every bridged call ........ - - * main/pbx.c, /: Merged revisions 136635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | - mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 - lines Don't allow Answer() to accept a negative argument. - Negative argument means an infinite delay and we don't want that. - ........ - - * main/channel.c, /: Merged revisions 136633 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | - mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 - lines Fix a calculation error I had made in the poll. The poll - would reset to 500 ms every time a non-voice frame was received. - The total time we poll should be 500 ms, so now we save the - amount of time left after the poll returned and use that as our - argument for the next call to poll ........ - - * main/channel.c, /: Merged revisions 136631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | - mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 - lines Scrap the 500 ms delay when Asterisk auto-answers a - channel. Instead, poll the channel until receiving a voice frame. - The cap on this poll is 500 ms. The optional delay is still - allowable in the Answer() application, but the delay has been - moved back to its original position, after the call to the - channel's answer callback. The poll for the voice frame will not - happen if a delay is specified when calling Answer(). (closes - issue #12708) Reported by: kactus ........ - -2008-08-07 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn_config.c, channels/chan_misdn.c, /, - configs/misdn.conf.sample: Merged revisions 136594 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 - (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) - | 5 lines * The allowed_bearers setting in misdn.conf misspelled - one of its options: digital_restricted. * Fixed some other - spelling errors and typos. ........ ................ - -2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/doxyref.h, /: Merged revisions 136542 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 - (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - ........ ................ - -2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) - | 15 lines Merged revisions 136488 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) - | 7 lines Update persistent state on all exit conditions. (closes - issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt - uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon - ........ ................ - -2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 - (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) - | 4 lines -C option takes a filename, not a directory path. - (closes issue #13007) Reported by: klaus3000 ........ - ................ - - * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) - | 7 lines Persist DIALGROUP() values in astdb (closes issue - #13138) Reported by: Corydon76 Patches: - 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) - Tested by: pj ........ - -2008-08-06 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com> - - * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 - (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug - 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame - type, there are places where ast_rtp_new_source may be called - where the tech_pvt of a channel may not yet have an rtp structure - allocated. This caused a crash in chan_skinny, which was fixed - earlier, but now the same crash has been reported against - chan_h323 as well. It seems that the best solution is to modify - ast_rtp_new_source to not attempt to set the marker bit if the - rtp structure passed in is NULL. This change to - ast_rtp_new_source also allows the removal of what is now a - redundant pointer check from chan_skinny. (closes issue #13247) - Reported by: pj ........ ................ - -2008-08-06 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net> - - * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | - oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - - Formatting - Changing debug messages from VERBOSE to DEBUG - channel - Adding a few todo's - Adding a few more "XMPP"'s to - compliment Jabber... ........ - -2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 135950 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) - | 12 lines Merged revisions 135949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) - | 4 lines Fix a longstanding bug in channel walking logic, and - fix the explanation to make sense. (Closes issue #13124) ........ - ................ - - * /, main/translate.c: Merged revisions 135938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) - | 12 lines Merged revisions 135915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) - | 4 lines Since powerof() can return an error condition, it's - foolhardy not to detect and deal with that condition. (Related to - issue #13240) ........ ................ - - * include/asterisk/threadstorage.h, include/asterisk/utils.h, /: - Merged revisions 135900 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) - | 12 lines Merged revisions 135899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) - | 4 lines 1) Bugfix for debugging code 2) Reduce compiler - warnings for another section of debugging code (Closes issue - #13237) ........ ................ - -2008-08-06 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/abstract_jb.h, main/channel.c, /, - main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions - 135851 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug - 2008) | 48 lines Merged revisions 135841,135847,135850 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug - 2008) | 27 lines Merging the issue11259 branch. The purpose of - this branch was to take into account "burps" which could cause - jitterbuffers to misbehave. One such example is if the L option - to Dial() were used to inject audio into a bridged conversation - at regular intervals. Since the audio here was not passed through - the jitterbuffer, it would cause a gap in the jitterbuffer's - timestamps which would cause a frames to be dropped for a brief - period. Now ast_generic_bridge will empty and reset the - jitterbuffer each time it is called. This causes injected audio - to be handled properly. ast_generic_bridge also will empty and - reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE - frame since the change in audio source could negatively affect - the jitterbuffer. All of this was made possible by adding a new - public API call to the abstract_jb called ast_jb_empty_and_reset. - (closes issue #11259) Reported by: plack Tested by: putnopvut - ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, - 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel - that occurred when I was testing for a memory leak ........ - r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug - 2008) | 3 lines Remove properties that should not be here - ........ ................ - -2008-08-05 23:52 +0000 [r135822] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, - include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | - 42 lines Merged revisions 135799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | - 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf - I discovered that also, in the previous bug fixes and changes, - the cdr.conf 'unanswered' option is not being obeyed, so I fixed - this. And, yes, there are two 'answer' times involved in this - scenario, and I would agree with you, that the first answer time - is the time that should appear in the CDR. (the second 'answer' - time is the time that the bridge was begun). I made the necessary - adjustments, recording the first answer time into the peer cdr, - and then using that to override the bridge cdr's value. To get - the 'unanswered' CDRs to appear, I purposely output them, using - the dial cmd to mark them as DIALED (with a new flag), and - outputting them if they bear that flag, and you are in the right - mode. I also corrected one small mention of the Zap device to - equally consider the dahdi device. I heavily tested 10-sec-wait - macros in dial, and without the macro call; I tested hangups - while the macro was running vs. letting the macro complete and - the bridge form. Looks OK. Removed all the instrumentation and - debug. ........ ................ - -2008-08-05 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 - (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) - | 9 lines In a conversion to use ast_strlen_zero, the meaning of - the flag IAX_HASCALLERID was perverted. This change reverts IAX2 - to the original meaning, which was, that the callerid set on the - client should be overridden on the server, even if that means the - resulting callerid is blank. In other words, if you set - "callerid=" in the IAX config, then the callerid should be - overridden to blank, even if set on the client. Note that there's - a distinction, even on realtime, between the field not existing - (NULL in databases) and the field existing, but set to blank - (override callerid to blank). ........ ................ - -2008-08-05 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com> - - * main/cli.c, /: Merged revisions 135598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug - 2008) | 9 lines Merged revisions 135597 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug - 2008) | 1 line Use PATH_MAX for filenames ........ - ................ - -2008-08-04 20:15 +0000 [r135538] Russell Bryant <russell@digium.com> - - * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135537 | russell | 2008-08-04 15:15:27 -0500 - (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) - | 2 lines fix a config sample typo ........ ................ - -2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.mandriva.asterisk (added), Makefile, - contrib/init.d/rc.mandrake.asterisk (removed), /, - contrib/init.d/rc.mandriva.zaptel (added), - contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions - 135485 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | - tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines - Rename Mandrake scripts to Mandriva (Closes issue #13221) - ........ - - * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 - (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) - | 2 lines Define ASTSBINDIR for script (Closes issue #13221) - ........ ................ - - * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 - (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) - | 6 lines Memory leak on unload (closes issue #13231) Reported - by: eliel Patches: app_voicemail.leak.patch uploaded by eliel - (license 64) ........ ................ - -2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com> - - * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135474 | russell | 2008-08-04 11:28:07 -0500 - (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) - | 2 lines Add a minor clarification to the documentation of - mohinterpret and mohsuggest ........ ................ - - * /, channels/chan_console.c: Merged revisions 135439 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) - | 4 lines Be explicit that we don't want a result from this - callback. The callback would never indicate a match, so nothing - would have been returned anyway, but it was still a poor example - of proper usage. ........ - -2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 135265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | - murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines - (closes issue #13202) Reported by: falves11 Tested by: murf - falves11 == The changes I introduce here seem to clear up the - problem for me. However, if they do not for you, please reopen - this bug, and we'll keep digging. The root of this problem seems - to be a subtle memory corruption introduced when creating an - extension with an empty extension name. While valgrind cannot - detect it outside of DEBUG_MALLOC mode, when compiled with - DEBUG_MALLOC, this is certain death. The code in main/features.c - is a puzzle to me. On the initial module load, the code is - attempting to add the parking extension before the features.conf - file has even been opened! I just wrapped the offending call with - an if() that will not try to add the extension if the extension - name is empty. THis seems to solve the corruption, and let the - "memory show allocations" work as one would expect. But, really, - adding an extension with an empty name is a seriously bad thing - to allow, as it will mess up all the pattern matching algorithms, - etc. So, I added a statement to the add_extension2 code to return - a -1 if this is attempted. in 1.6.0, the changes to only - main/pbx.c were applicable, as apparently the code added to - main/features by jpeeler were not included in 1.6.0. ........ - -2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com> - - * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug - 2008) | 6 lines Remove some code that used to do something but - does not anymore, mainly to get rid of a shadow warning (but this - seemed legitimate enough to fix here instead of in my branch). - Thanks to putnopvut for taking a look as well. ........ - -2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | - tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines - Picky, picky, buildbot ........ - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 135126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | - tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines - SIP should use the transport type set in the Moved Temporarily - for the next invite. (closes issue #11843) Reported by: - pestermann Patches: - 20080723__issue11843_302_ignores_transport_16branch.diff uploaded - by bbryant (license 36) - 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by - bbryant (license 36) Tested by: pabelanger ........ - -2008-08-01 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged - revisions 135067-135068 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | - mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 - lines IMAP storage functioned under the assumption that folders - such as "Work" and "Family" would be subfolders of the INBOX. - This is an invalid assumption to make, but it could be desirable - to set up folders in this manner, so a new option for - voicemail.conf, "imapparentfolder" has been added to allow for - this. (closes issue #13142) Reported by: jaroth Patches: - parentfolder.patch uploaded by jaroth (license 50) ........ - r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug - 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE - defines... ........ - -2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info> - - * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) - | 10 lines Merged revisions 135058 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) - | 2 lines make app_ices compile on OpenBSD. ........ - ................ - - * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 - (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) - | 8 lines fix some potential deadlocks in chan_skinny (closes - issue #13215) Reported by: qwell Patches: - 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) - Tested by: mvanbaak ........ ................ - -2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/http.c: Merged revisions 135016 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul - 2008) | 11 lines Merged revisions 134983 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul - 2008) | 3 lines accomodate users who seem to lack a sense of - humor :-) ........ ................ - -2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com> - - * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions - 134980 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) - | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 - | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 - lines Specify codecs in callfiles and manager, to allow video - calls to be set up from callfiles and AMI. (closes issue #9531) - Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt - uploaded by Corydon76 (license 14) - 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license - 14) Tested by: Corydon76 ........ ................ - - * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) - | 2 lines Switch command order, to meet with current specs - ........ - -2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 134922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | - 63 lines Merged revisions 134883 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | - 51 lines (closes issue #11849) Reported by: greyvoip Tested by: - murf OK, a few days of debugging, a bunch of instrumentation in - chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook - pages of notes later, I have made the small tweek necc. to get - the start time right on the second CDR when: A Calls B B answ. A - hits Xfer button on sip phone, A dials C and hits the OK button, - A hangs up C answers ringing phone B and C converse B and/or C - hangs up But does not harm the scenario where: A Calls B B answ. - B hits xfer button on sip phone, B dials C and hits the OK - button, B hangs up C answers ringing phone A and C converse A - and/or C hangs up The difference in start times on the second CDR - is because of a Masquerade on the B channel when the xfer number - is sent. It ends up replacing the CDR on the B channel with a - duplicate, which ends up getting tossed out. We keep a pointer to - the first CDR, and update *that* after the bridge closes. But, - only if the CDR has changed. I hope this change is specific - enough not to muck up any current CDR-based apps. In my defence, - I assert that the previous information was wrong, and this change - fixes it, and possibly other similar scenarios. I wonder if I - should be doing the same thing for the channel, as I did for the - peer, but I can't think of a scenario this might affect. I leave - it, then, as an exersize for the users, to find the scenario - where the chan's CDR changes and loses the proper start time. - ........ ................ - -2008-07-31 19:41 +0000 [r134918] Russell Bryant <russell@digium.com> - - * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) - | 17 lines Merged revisions 134915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) - | 9 lines Get app_ices working again (closes issue #12981) - Reported by: dlogan Patches: - 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant - (license 36) 20080709__app_ices_v2_update_14.diff uploaded by - bbryant (license 36) Tested by: bbryant ........ ................ - -2008-07-31 16:53 +0000 [r134816] Russell Bryant <russell@digium.com> - - * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) - | 15 lines Merged revisions 134814 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) - | 7 lines In case we have some processing threads that free more - frames than they allocate, do not let the frame cache grow - forever. (closes issue #13160) Reported by: tavius Tested by: - tavius, russell ........ ................ - -2008-07-31 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul - 2008) | 24 lines Merged revisions 134758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul - 2008) | 16 lines Add more timeout checks into app_queue, - specifically targeting areas where an unknown and potentially - long time has just elapsed. Also added a check to try_calling() - to return early if the timeout has elapsed instead of potentially - setting a negative timeout for the call (thus making it have *no* - timeout at all). (closes issue #13186) Reported by: - miquel_cabrespina Patches: 13186.diff uploaded by putnopvut - (license 60) Tested by: miquel_cabrespina ........ - ................ - -2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com> - - * main/sched.c, /, include/asterisk/sched.h: Merged revisions - 134703 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | - tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines - Oops, wrong define ........ - - * /, configure, configure.ac: Merged revisions 134650 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 - (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) - | 4 lines Qwell pointed out, via IRC, that the previous fix only - worked when explicitly set. When nothing is set, and the option - is implied, it breaks, because configure sets the prefix to - 'NONE'. Fixing. ........ ................ - -2008-07-30 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul - 2008) | 15 lines Merged revisions 134556 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | - mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 - lines Fix the parsing of the "reason" parameter in the Diversion: - header. (closes issue #13195) Reported by: woodsfsg ........ - ................ - -2008-07-30 20:39 +0000 [r134597] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) - | 14 lines Merged revisions 134595 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) - | 6 lines Reduce stack consumption by 12.5% of the max stack size - to fix a crash when compiled with LOW_MEMORY. (closes issue - #13154) Reported by: edantie ........ ................ - -2008-07-30 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | - mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 - lines Fix the parsing of the "reason" parameter in the Diversion: - header. (closes issue #13195) Reported by: woodsfsg ........ - -2008-07-30 19:56 +0000 [r134542] Russell Bryant <russell@digium.com> - - * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) - | 12 lines Merged revisions 134540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) - | 4 lines Fix a memory leak in func_curl. Every thread that used - this function leaked an allocation the size of a pointer. - (reported by jmls in #asterisk-dev) ........ ................ - -2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com> - - * /, configure, configure.ac: Merged revisions 134538 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 - (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) - | 4 lines Only override sysconfdir and mandir when prefix=/usr - (closes issue #13093) Reported by: pabelanger ........ - ................ - - * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | - tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines - Let "roundrobin" also reference rrmemory, for the 1.6 release (as - described in UPGRADE-1.4.txt) (Closes issue #13181) ........ - - * /, res/res_agi.c: Merged revisions 134481 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) - | 13 lines Merged revisions 134480 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) - | 5 lines launch_netscript sometimes returns -1, which fails to - set AGISTATUS. Map failure to -1, so that AGISTATUS is always - set. (closes issue #13199) Reported by: smw1218 ........ - ................ - -2008-07-30 18:33 +0000 [r134477] Mark Michelson <mmichelson@digium.com> - - * /, main/app.c: Merged revisions 134476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul - 2008) | 12 lines Merged revisions 134475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul - 2008) | 4 lines Fix a spot where a function could return without - bringing a channel out of autoservice. ........ ................ - -2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 134355 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul - 2008) | 10 lines Merged revisions 134352 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul - 2008) | 2 lines use the proper method for building version.h - ........ ................ - -2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, - apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: - Merged revisions 134260 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | - kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 - lines build against the now-typedef-free dahdi/user.h, and remove - some #ifdefs for features that will always be present in DAHDI - ........ - -2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 - (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) - | 7 lines Detect when sox fails to raise the volume, because sox - can't read the file. (closes issue #12939) Reported by: - rickbradley Patches: 20080728__bug12939.diff.txt uploaded by - Corydon76 (license 14) Tested by: rickbradley ........ - ................ - -2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com> - - * /, configure, main/Makefile, configure.ac, CHANGES: Merged - revisions 134125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | - mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 - lines This commit compensates for buggy poll(2) implementations. - Asterisk has, for a long time, had its own implementation of - poll(2) which just used the input arguments to call select(2). In - 1.4, this internal implementation was used for Darwin systems. - This was removed in Asterisk trunk at some point, but it seems as - though this was not the right move to make. On Mac OS X, it - appears as though the poll used to gather CLI input does not - respond properly when connecting via a remote Asterisk console. - Reverting to the use of Asterisk's poll fixed the issue. Also, - there is now an option for the configure script, - --enable-internal-poll, which will allow for anyone to use - Asterisk's internal poll implementation in case they suspect that - their system's poll implementation is buggy. closes issue #11928) - Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded - by putnopvut (license 60) ........ - -2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_parkandannounce.c, main/loader.c, sample.call, - contrib/scripts/autosupport, build_tools/cflags.xml, - main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, - configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, - doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions - 134086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | - kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 - lines remove remaining Zaptel references in various places - ........ - -2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, - /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions - 134050 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | - mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 - lines merging the zap_and_dahdi_trunk branch up to trunk ........ - -2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com> - - * main/asterisk.c, include/asterisk/doxyref.h, /: Include the - licensing page in 1.6.0 as well. Now, this page exists in 1.4, - trunk, and 1.6.0. - - * /: unblock 133575 - - * /, main/devicestate.c: Merged revisions 133945-133946 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 - Jul 2008) | 6 lines ast_device_state() gets called in two - different ways. The first way is when called from elsewhere in - Asterisk to find the current state of a device. In that case, we - want to use the cached value if it exists. The other way is when - processing a device state change. In that case, we do not want to - check the cache because returning the last known state is counter - productive. ........ r133946 | russell | 2008-07-26 10:16:20 - -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache - argument ........ - -2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 - Jul 2008) | 6 lines Update version (closes issue #13163) Reported - by: suretec Patches: asterisk.ldif uploaded by suretec (license - 70) ........ - -2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com> - - * /: Blocking revert of code changes in r133770 - - * main/http.c: Include the http_decode function from trunk to - replace the + with a space. - -2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com> - - * /: Blocking a fix from trunk for the function http_decode. 1.6.0 - does not have this function. - -2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: - Merged revisions 133665 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) - | 16 lines Merged revisions 133649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) - | 8 lines Fix some errant device states by making the devicestate - API more strict in terms of the device argument (only without the - unique identifier appended). (closes issue #12771) Reported by: - davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 - (license 14) Tested by: davidw, jvandal, murf ........ - ................ - -2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com> - - * /, LICENSE: Merged revisions 133579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) - | 18 lines Merged revisions 133578 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r133578 | russell | 2008-07-25 10:00:31 -0500 - (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) - | 2 lines Fix the IAX2 URI for calling Digium ........ - ................ ................ - -2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul - 2008) | 15 lines Merged revisions 133572 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul - 2008) | 7 lines We need to make sure to null-terminate the "name" - portion of SIP URI parameters so that there are no bogus - comparisons. Thanks to bbryant for pointing this out. ........ - ................ - -2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | - russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines - Minor coding guidelines tweaks ... - Use ast_strlen_zero in one - place - check for successful string comparison the way most of - Asterisk code does it ........ - -2008-07-24 21:31 +0000 [r133524] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) - | 11 lines Merged revisions 133488 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) - | 3 lines Fix rtautoclear and rtcachefriends (Closes issue - #12707) ........ ................ - -2008-07-24 20:41 +0000 [r133487] Russell Bryant <russell@digium.com> - - * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) - | 3 lines I made this change from DEVICE_STATE to - DEVICE_STATE_CHANGE, but I had it backwards, this is the right - event to subscribe to ... ........ - -2008-07-24 19:55 +0000 [r133449] Mark Michelson <mmichelson@digium.com> - - * /, main/logger.c: Merged revisions 133448 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | - mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 - lines Print the correct PID in log messages. Prior to this - commit, only the logger thread's PID would be printed. (closes - issue #13150) Reported by: atis Patches: log_pid.diff uploaded by - putnopvut (license 60) Tested by: eliel ........ - -2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions - 133400 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | - tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines - Build the logrotate script according to paths (Closes issue - #13147) ........ - - * Makefile, /: Merged revisions 133391 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | - tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines - Optionally install logrotate file (Closes issue #13148) ........ - -2008-07-23 22:07 +0000 [r133300] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 133299 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | - murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines - (closes issue #13144) Reported by: murf Tested by: murf For: J. - Geis The 'data' field in the ast_exten struct was being 'moved' - from the current dialplan to the replacement dialplan. This was - not good, as the current dialplan could have problems in the time - between the change and when the new dialplan is swapped in. So, I - modified the merge_and_delete code to strdup the 'data' field - (the args to the app call), and then it's freed as normal. I - improved a few messages; I added code to limit the number of - calls to the context_merge_incls_swits_igps_other_registrars() to - one per context. I don't think having it called multiple times - per context was doing anything bad, but it was inefficient. I - hope this fixes the problems Mr. Geiss was noting in - asterisk-users, see - http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html - ........ - -2008-07-23 21:50 +0000 [r133297] Jason Parker <jparker@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 - (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul - 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect - ........ ................ - -2008-07-23 20:39 +0000 [r133218] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | - bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines - Fix issue where tcp in sip is enabled by default, despite what it - says in the config sample file. Also fix "sip show settings" for - tcp connections. ........ - -2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, - /: Merged revisions 133171 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul - 2008) | 20 lines Merged revisions 133169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul - 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN - in app_chanspy should be set at load time, not at compile time, - since dahdi_chan_name is determined at load time. Also changed - the next_unique_id_to_use to have the static qualifier. Also - added the dahdi_chan_name_len variable so that - strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for - the suggestion. ........ ................ - - * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul - 2008) | 13 lines Merged revisions 133104 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul - 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is - twelve. The strncmp call in next_channel should account for this. - ........ ................ - - * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul - 2008) | 14 lines Merged revisions 133101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul - 2008) | 6 lines Update the "last" channel in next_channel in - app_chanspy so that the same pseudo channel isn't constantly - returned. related to issue #13124 ........ ................ - - * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 - (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul - 2008) | 7 lines Small cleanup. Move the declaration of the - DAHDI_SPANINFO variable to the block where it is used. This - allows one less #ifdef HAVE_PRI to clutter things up. Thanks to - Tzafrir for pointing this out on #asterisk-dev ........ - ................ - -2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) - | 6 lines Yet another conversion of '|' to ',' (closes issue - #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch - uploaded by eliel (license 64) ........ - - * contrib/scripts/asterisk.logrotate (added), /: Merged revisions - 132977 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | - tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines - Add logrotate script for Asterisk (closes issue #13085) Reported - by: pabelanger Patches: logrotate uploaded by pabelanger (license - 224) ........ - -2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming <kpfleming@digium.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 - (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 - Jul 2008) | 1 line another Fix because of r119585, this commit - has broken high frequented BRI Ports, there was a possibility - that a channel, that was marked as in_use would be reused later, - the corresponding port could got stuck then. So it is recommended - to upgrade for chan_misdn users. ........ ................ - r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul - 2008) | 2 lines use correct function name... please compile with - --enable-dev-mode ................ - - * include/asterisk/stringfields.h, /, main/utils.c: Merged - revisions 132964 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul - 2008) | 10 lines Merged revisions 132872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul - 2008) | 2 lines minor optimization for stringfields: when a field - is being set to a larger value than it currently contains and it - happens to be the most recent field allocated from the currentl - pool, it is possible to 'grow' it without having to waste the - space it is currently using (or potentially even allocate a new - pool) ........ ................ - -2008-07-23 08:18 +0000 [r132824] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | - oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines - Well, the content of a channel variable may be longer than the - size of a pointer... Thanks, eliel! Reported by: eliel Patches: - chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) - (closes issue #13135) ........ - -2008-07-22 22:20 +0000 [r132797] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul - 2008) | 11 lines Merged revisions 132777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ Allow - Spiraled INVITEs to work correctly within Asterisk. Prior to this - change, a spiraled INVITE would cause a 482 Loop Detected to be - sent to the caller. With this change, if a potential loop is - detected, the Request-URI is inspected to see if it has changed - from what was originally received. If pedantic mode is on, then - this inspection is fully RFC 3261 compliant. If pedantic mode is - not on, then a string comparison is used to test the equality of - the two R-URIs. This has been tested by using OpenSER to rewrite - the R-URI and send the INVITE back to Asterisk. (closes issue - #7403) Reported by: stephen_dredge Modified: - branches/1.4/channels/chan_sip.c ........ ................ - -2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul - 2008) | 2 lines correct fix made in r132777... the code *did* - compile in dev-mode, as long as libpri was installed and enabled - ........ - -2008-07-22 21:59 +0000 [r132782] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged - revisions 132703 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 - lines Merged revisions 132645 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 - lines The most common question on the #asterisk iRC channel and - on mailing lists seems to be in regards to an error message when - retransmit fails. This is frequently misunderstood as a failure - of Asterisk, not a failure of the network to reach the other - party. This document tries to assist the Asterisk user in sorting - out these issues by explaining the logic and pointing at some - possible causes. Hopefully, we will get other questions now :-) - ........ ................ - -2008-07-22 21:55 +0000 [r132780] Tilghman Lesher <tlesher@digium.com> - - * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged - revisions 132778 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) - | 18 lines Merged revisions 132713 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 - (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) - | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ - ................ ................ - -2008-07-22 21:54 +0000 [r132779] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul - 2008) | 3 lines Get chan_dahdi to compile in devmode ........ - -2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 - (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul - 2008) | 6 lines ensure that if any alarms exist at channel - creation time, they are handled identically to if they occurred - later, so that later alarm clearing will work properly and 'make - sense' (closes issue #12160) Reported by: tzafrir ........ - ................ - - * /, configure, configure.ac, acinclude.m4: Merged revisions 132705 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 - (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul - 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' - description of what it is doing ........ ................ - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, - configure, include/asterisk/autoconfig.h.in, configure.ac: Merged - revisions 132643 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul - 2008) | 10 lines Merged revisions 132641 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul - 2008) | 2 lines use renamed libpri API call for controlling this - feature (was improperly named before) ........ ................ - - * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 - (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul - 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI - spans, and don't attempt to use channel 24 as a D-channel on - spans of unexpected sizes ........ ................ - -2008-07-21 21:13 +0000 [r132515] Brett Bryant <bbryant@digium.com> - - * configs/features.conf.sample, configs/gtalk.conf.sample, /, - configs/jingle.conf.sample, configs/manager.conf.sample: Merged - revisions 132514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | - bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines - Update configuration files to add missing options for jingle, - gtalk, manager.conf, and features.conf. (closes issue #13128) - Reported by: caio1982 Patches: missing_options1.diff uploaded by - caio1982 (license 22) ........ - -2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher <tlesher@digium.com> - - * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): - Merged revisions 132511 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | - tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines - (Step 2 of 2) ........ - - * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h - (added), build_tools/cflags.xml, main/fskmodem_float.c (added), - /, main/tdd.c, include/asterisk/fskmodem.h (removed), - main/fskmodem_int.c (added), main/callerid.c, - include/asterisk/fskmodem_float.h (added): Merged revisions - 132510 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | - tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines - Optionally build integer-based routines for FSK tone decoding - (but default to the more accurate float-based routines). (Closes - issue #11679) (Step 1 of 2) ........ - -2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant <bbryant@digium.com> - - * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | - bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines - Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't - supported on a channel (yet _another_ useful patch by eliel). - (closes issue #13081) Reported by: eliel Patches: - app_sendtext.c.patch uploaded by eliel (license 64) Tested by: - eliel ........ - - * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | - bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines - Fix bug where ast_parse_arg would inadvertantly enable sip tcp - when parsing a tcpbindaddr if it was disabled. (closes issue - #13117) Reported by: pj ........ - - * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) - | 3 lines Fix an issue in iax2 where a call that's been rejected - still kept an open channel on the side that attempted to make the - call (not the side of the call that rejected the call). Changes - were load tested and also approved by Russell. ........ - -2008-07-21 15:34 +0000 [r132426] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) - | 2 lines make buffers config option (chan_dahdi.conf) parsing - safer and added logging in case of failure ........ - -2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant <russell@digium.com> - - * apps/app_jack.c, include/asterisk/libresample.h (removed), /, - build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, main/Makefile, main/libresample - (removed), configure.ac, codecs/codec_resample.c, makeopts.in: - Merged revisions 132390 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | - russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines - Remove libresample from the Asterisk source tree. It is now - available in its own repository, and must be installed like any - other library for Asterisk to use. The two modules that require - it are codec_resample and app_jack. To install libresample: $ svn - co http://svn.digium.com/svn/libresample/trunk libresample $ cd - libresample $ ./configure $ make $ sudo make install This code is - currently in our own repository because the build system did not - include the appropriate targets for building a dynamic library or - for installing the library. ........ - - * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions - 132388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | - russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines - Enable higher quality resampling, as it doesn't have a noticeable - performance impact on my machine .. ........ - -2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming <kpfleming@digium.com> - - * /, LICENSE: Merged revisions 132312 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul - 2008) | 10 lines Merged revisions 132311 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul - 2008) | 2 lines grant a license exception to allow distribution - of Asterisk binaries that use the UW IMAP Toolkit (which is - licensed under a non-GPL-compatible license) ........ - ................ - -2008-07-19 10:47 +0000 [r132278] Michiel van Baak <michiel@vanbaak.info> - - * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) - | 7 lines fix a couple of comments in sqlite resource driver. - (closes issue #13110) Reported by: gknispel_proformatique - Patches: res_config_sqlite_comments.patch uploaded by gknispel - (license 261) ........ - -2008-07-18 22:20 +0000 [r132245] Brett Bryant <bbryant@digium.com> - - * main/manager.c, /: Merged revisions 132242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | - bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines - Fixes problem where manager users loaded from users.conf would be - removed early (before the routine to load the configuration was - finished) because a variable wasn't initialized. ........ - -2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c: Merged revisions 132113 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) - | 14 lines Merged revisions 132112 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) - | 6 lines Fix for Taiwanese number syntax (closes issue #12319) - Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch - uploaded by CharlesWang (license 444) ........ ................ - -2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | - 1 line Make sure we break the poll so that messages queued will - be sent on the SS7 when using CLI commands for blocking and - blocking of CICs and linksets. ........ - -2008-07-18 18:51 +0000 [r132110] Tilghman Lesher <tlesher@digium.com> - - * main/config.c, /: Merged revisions 132109 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) - | 14 lines Merged revisions 132107 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) - | 6 lines Textual clarification (closes issue #13106) Reported - by: flefoll Patches: config.c.br14.120173.patch-unknown-directive - uploaded by flefoll (license 244) ........ ................ - -2008-07-18 17:56 +0000 [r132051] Brett Bryant <bbryant@digium.com> - - * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 - Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and - change cdr_radius.c to use the same keyword as the other files - (patch by eliel). (closes issue #13104) Reported by: eliel - Patches: revision.patch uploaded by eliel (license 64) ........ - -2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher <tlesher@digium.com> - - * main/sched.c, /: Merged revisions 131989 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) - | 10 lines Merged revisions 131988 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) - | 2 lines Oops ........ ................ - - * main/sched.c, /, include/asterisk/sched.h: Merged revisions - 131986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) - | 10 lines Merged revisions 131985 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) - | 2 lines Preserve ABI compatibility with last change ........ - ................ - - * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: - Merged revisions 131982 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) - | 10 lines Merged revisions 131970 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) - | 2 lines Make the ast_assert call within ast_sched_del report - something useful. ........ ................ - -2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming <kpfleming@digium.com> - - * main/dlfcn.c (removed), main/loader.c, /, main/Makefile, - include/asterisk/dlfcn-compat.h (removed): Merged revisions - 131923 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul - 2008) | 10 lines Merged revisions 131921 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul - 2008) | 2 lines remove the dlfcn compatibility stuff, because no - platforms that Asterisk currently runs on it use it, and it - doesn't build anyway ........ ................ - -2008-07-18 15:39 +0000 [r131917] Brett Bryant <bbryant@digium.com> - - * /, main/features.c: Merged revisions 131916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) - | 12 lines Merged revisions 131915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) - | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER - variable isn't always set to the other end of the blind transfer. - (closes issue #12586) ........ ................ - -2008-07-17 22:45 +0000 [r131869] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) - | 6 lines Add configuration option to chan_dahdi.conf to allow - buffering policy and number of buffers to be configured per - channel. Syntax: buffers=<num of buffers>,<policy> Where the - number of buffers is some non-negative integer and the policy is - either "full", "half", or "immediate". ........ - -2008-07-17 21:27 +0000 [r131830] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 | - mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 - lines Document that the duration of dtmf may be passed to the - SendDTMF application. Also correct the default pause between - digits. (closes issue #13102) Reported by: eliel Patches: - app_senddtmf.c.patch uploaded by eliel (license 64) ........ - -2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500 - (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) - | 7 lines Revert part of issue #5620 (revision 6965) as it - appears that it was in error. This should fix talk call progress - on analog lines. (closes issue #12178) Reported by: michael-fig - Patches: 20080717__bug12178.diff.txt uploaded by Corydon76 - (license 14) ........ ................ - - * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) - | 6 lines Fix memory leaks (closes issue #13099) Reported by: - gknispel_proformatique Patches: - res_config_sqlite_leak_on_error.patch uploaded by gknispel - (license 261) ........ - -2008-07-17 18:15 +0000 [r131718] Brett Bryant <bbryant@digium.com> - - * /, main/features.c: Merged revisions 131717 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 | - bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines - Fix a memory leak in register_group_feature when attempting to - register a feature without specifying a group or feature to - register. (closes issue #13101) Reported by: eliel Patches: - features.c.patch uploaded by eliel (license 64) ........ - -2008-07-17 15:46 +0000 [r131682] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) - | 4 lines Fix memory leak. (Closes issue #13096) Reported by - gknispel_proformatique ........ - -2008-07-16 23:56 +0000 [r131571] Steve Murphy <murf@digium.com> - - * /: The commit from 131570 should not be applied to 1.6.0, as it - is not as necessary, because log_show_lock in trunk is not - available in 1.6.0, and is estimated to be the only function that - might care about the lock_addr values. - -2008-07-16 22:18 +0000 [r131493] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500 - (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) - | 6 lines Fix a bug in iax2 registration that allowed peers to - register with case-insensitive names (user_cmp_cb and peer_cmp_cb - are now both case-sensitive). (closes issue #13091) ........ - ................ - -2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant <bbryant@digium.com> - - * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) - | 4 lines Fixes sysinfo operator issue also fixed elsewhere in - r131445. (issue #13057) ........ - - * main/asterisk.c, /: Merged revisions 131445 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 | - bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines - Fixes an issue with "core show sysinfo" that used the wrong - operator to calculate the number of bytes from a sysinfo - structure. unsigned long. (closes issue #13057) Reported by: - eliel Patches: asterisk.c.patch uploaded by eliel (license 64) - ........ - -2008-07-16 20:48 +0000 [r131423] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r131422 | russell | 2008-07-16 15:48:27 -0500 - (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) - | 7 lines Always ensure that the channel's tech_pvt reference is - NULL after calling the destroy callback. (closes issue #13060) - Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch - uploaded by jpgrayson (license 492) ........ ................ - -2008-07-16 20:24 +0000 [r131301-131378] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul - 2008) | 22 lines Merged revisions 131369 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul - 2008) | 14 lines Move the init_queue call back to where it used - to be (changed Sept 12 last year). It was moved then to prevent a - memory leak. Since then, the same memory leak recurred and was - fixed in a better way. Now it has been found that the placement - of this init_queue call can cause problems if a realtime queue - has values changed to an empty string. The problem is that the - default value for that queue parameter would not be set. (closes - issue #13084) Reported by: elbriga ........ ................ - - * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul - 2008) | 9 lines Don't try to dereference the dbfile pointer if we - know that it's NULL. (closes issue #13092) Reported by: - gknispel_proformatique Patches: - trunk_sqlite_check_vars_null.patch uploaded by gknispel (license - 261) ........ - - * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul - 2008) | 14 lines Merged revisions 131357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul - 2008) | 6 lines Apparently, "thread safety" is important, - whatever that means. :P (Thanks Russell!) ........ - ................ - - * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul - 2008) | 21 lines Merged revisions 131299 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul - 2008) | 13 lines Make absolutely certain that the transfer - datastore is removed from the calling channel once the caller is - finished in the queue. This could have weird con- sequences when - dialing local queue members when multiple transfers occur on a - single call. Also fixed a memory leak that would occur when an - attended transfer occurred from a queue member. (closes issue - #13047) Reported by: festr ........ ................ - -2008-07-16 18:20 +0000 [r131248] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | - 27 lines Merged revisions 131242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | - 19 lines (closes issue #13090) Reported by: murf The problem was - that, esoteric as it is, because the hangerupper context - immediately preceded the std-priv-extent macro, that the checking - code accidentally would fall from traversing hangerupper into the - std-priv-exten macro, where it would hit the hangerupper in the - 'includes', and proceed into an infinite recursion. A small fix - to traverse into the statements of the context instead of the - context solves this issue. I also added some commented out - printfs for debug, which were pretty handy in the face of a dorky - gdb. This was a problem around since the package was first - written; but evidently pretty rare in turning up in the field. - ........ ................ - -2008-07-16 15:04 +0000 [r131206] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_agent.c: add missing terminator argument for - ast_event_subscribe(). Without it the function will randomly walk - on the stack possibly causing a panic - -2008-07-16 00:54 +0000 [r131168] Tilghman Lesher <tlesher@digium.com> - - * /, main/logger.c: Merged revisions 131166 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 | - tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines - Fix rotate strategy (Closes issue #13086) ........ - -2008-07-15 23:41 +0000 [r131131] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 131129 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 | - murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines - (closes issue #12960) Reported by: mnicholson Spent most of the - day on this bug, and the solution was so simple. Just had to find - and understand the problem. The problem was, that the routine to - copy the existing switches, includes, and ignorepats from the old - context to the new one, wasn't getting called when the context is - already existent. (In other words, if AEL is adding a new context - to the mix, they get copied, but if pbx_config already defined a - context, then the copy wasn't happening. This made no sense, so I - moved the call to copy the includes & etc, no matter the case. - ........ - -2008-07-15 18:47 +0000 [r131073] Russell Bryant <russell@digium.com> - - * /, res/res_agi.c: Merged revisions 131072 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 | - russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines - Fix a couple of places in res_agi where the agi_commands lock - would not be released, causing a deadlock. (Reported by mvanbaak - in #asterisk-dev, discovered by bbryant's change to the lock - tracking code to yell at you if a thread exits with a lock still - held) ........ - -2008-07-15 18:29 +0000 [r131060] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged - revisions 131044 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) - | 16 lines Merged revisions 130959 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) - | 8 lines astman_send_error does not need a newline appended -- - the API takes care of that for us. (closes issue #13068) Reported - by: gknispel_proformatique Patches: - asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) - asterisk_trunk_astman_send.patch uploaded by gknispel (license - 261) ........ ................ - -2008-07-15 18:00 +0000 [r131014] Michiel van Baak <michiel@vanbaak.info> - - * main/cdr.c, /: Merged revisions 131013 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) - | 15 lines Merged revisions 131012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) - | 7 lines remove 4 lines of redundant code. (closes issue #13080) - Reported by: gknispel_proformatique Patches: - trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) - ........ ................ - -2008-07-15 13:14 +0000 [r130946] Steve Murphy <murf@digium.com> - - * utils/conf2ael.c, utils/Makefile, res/ael/pval.c, - channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c, - pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, - utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c, - res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c, - apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /, - channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, - channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - Merging this rev from trunk to 1.6.0 was not simple. Why? Because - we've enhanced trunk to do a [fast] merge-and-delete operation - which also solved problems with contexts having entries from - different registrars. Fast as in the amount of time the contexts - are locked down. That *is* fast, but traversing the entire - dialplan looking for priorities to delete takes more time - overall. This particular fix involved pulling in those - enhancements from trunk, along with all the various fixes and - refinements made along the way. Merging all this from trunk into - 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert - all but the prop changes c. catalog all revisions to pbx.c since - 1.6.0 was forked (at rev 105596). d. catalog all revisions to - pbx.c in trunk since 1.6.0 was forked, making special note of all - revs that were not merged into 1.6.0. e. study each rev in trunk - not applied to 1.6.0, and determine if it was involved in the - merge_and_delete enhancements in trunk. 25 commits were done in - 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 - additional changes, of which 7 were involved in the - merge_and_delete enhancements: 106757 108894 109169 116461 123358 - 130145 130297 f. Go to trunk and collect patches, one by one, of - the changes made by each rev across the entire source tree, using - svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0, - and resolve all failures and compilation problems before - proceding to the next patch. h. test the stuff. i. profit! - ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul - 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested - by: murf (closes issue #12960) Reported by: mnicholson In this - 'omnibus' fix, I **think** I solved both the problem in 13041, - where unloading pbx_ael.so caused crashes, or incomplete removal - of previous registrar'ed entries. And I added code to completely - remove all includes, switches, and ignorepats that had a matching - registrar entry, which should appease 12960. I also added a lot - of seemingly useless brackets around single statement if's, which - helped debug so much that I'm leaving them there. I added a - routine to check the correlation between the extension tree lists - and the hashtab tables. It can be amazingly helpful when you have - lots of dialplan stuff, and need to narrow down where a problem - is occurring. It's ifdef'd out by default. I cleaned up the code - around the new CIDmatch code. It was leaving hanging extens with - bad ptrs, getting confused over which objects to remove, etc. I - tightened up the code and changed the call to remove_exten in the - merge_and_delete code. I added more conditions to check for empty - context worthy of deletion. It's not empty if there are any - includes, switches, or ignorepats present. If I've missed - anything, please re-open this bug, and be prepared to supply - example dialplan code. ........ - -2008-07-15 00:00 +0000 [r130891] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500 - (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) - | 8 lines Override the callerid in all cases when the callerid is - set in the user, not just when a remote callerid is set. Also, if - not set in the user, allow the remote CallerID to pass through. - (closes issue #12875) Reported by: dimas Patches: - 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) - ........ ................ - -2008-07-14 22:24 +0000 [r130795-130855] Mark Michelson <mmichelson@digium.com> - - * main/asterisk.c, /: Merged revisions 130854 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 | - mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 - lines Fix a memory leak in the case that /dev/null cannot be - opened when running startup commands from cli.conf (closes issue - #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by - eliel (license 64) ........ - - * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul - 2008) | 16 lines Merged revisions 130792 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul - 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in - app_dial to be sure there are no audiohooks present on the - channels involved. This fixed a one-way audio situation I had in - my test setup. I couldn't find any open issues that suggested - one-way audio with regards to mixmonitor (or other audiohook) - usage, though. ........ ................ - -2008-07-14 17:22 +0000 [r130752] Michiel van Baak <michiel@vanbaak.info> - - * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) - | 18 lines Merged revisions 130735 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) - | 10 lines notify the user that dnsmgr refresh wont work when - dnsmgr is not enabled. Previously this command would - automagically appear and disappear. This was confusing. (closes - issue #12796) Reported by: chappell Patches: - dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: - russell, chappell, mvanbaak ........ ................ - -2008-07-14 10:40 +0000 [r130636-130637] Russell Bryant <russell@digium.com> - - * /, include/asterisk/astobj.h: Merged revisions 129987 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129987 | russell | 2008-07-11 09:22:44 -0500 - (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) - | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ - ................ - - * /, main/audiohook.c: Merged revisions 130635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) - | 10 lines Merged revisions 130634 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) - | 2 lines Bump up the debug level for a message. ........ - ................ - -2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak <michiel@vanbaak.info> - - * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: - Merged revisions 130578 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 | - mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 - lines Make all sed calls Posix sed compatible. To make sure - nobody commits script-modified files we first make a backup of - asterisk.tex, run the script, generate the pdf and / or html, and - put the original asterisk.tex back. This will guard us for the - stuff that happened before that someone committed a locally - modified asterisk.tex, with changes done by this script. (closes - issue #13062) Reported by: mvanbaak Patches: - sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by: - mvanbaak Feedback from Corydon. Thanks for taking the time to go - through this. ........ - - * main/manager.c, /: Merged revisions 130574 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) - | 16 lines Merged revisions 130573 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) - | 8 lines fix memory leak when originate from manager cannot - create a thread (closes issue #13069) Reported by: - gknispel_proformatique Patches: - asterisk_trunk_action_originate.patch uploaded by gknispel - (license 261) Tested by: gknispel_proformatique, mvanbaak - ........ ................ - -2008-07-13 17:59 +0000 [r130516] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500 - (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) - | 4 lines Reverting 2 changesets, as it breaks incoming IAX2 - calls (Related to issue #12963) Reported by: mvanbaak ........ - ................ - -2008-07-13 15:00 +0000 [r130480] Michiel van Baak <michiel@vanbaak.info> - - * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) - | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This - got lost in commit 97634 ........ - -2008-07-13 02:35 +0000 [r130445] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) - | 2 lines Unlock list before returning ........ - -2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | - 1 line Support new TRANSPORT definitions in libss7 ........ - -2008-07-11 20:04 +0000 [r130238] Mark Michelson <mmichelson@digium.com> - - * /, main/audiohook.c: Merged revisions 130237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul - 2008) | 11 lines Merged revisions 130236 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul - 2008) | 3 lines Remove redundant logic ........ ................ - -2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c: - Merged revisions 130230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 | - tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines - Fix trunk breakage ........ - -2008-07-11 19:14 +0000 [r130175] Mark Michelson <mmichelson@digium.com> - - * /, main/audiohook.c: Merged revisions 130174 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul - 2008) | 15 lines Merged revisions 130173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul - 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While - this change has not been proven to fix any specific issue, it is - incorrect and could cause unforeseen problems. ........ - ................ - -2008-07-11 18:53 +0000 [r130171] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500 - (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) - | 7 lines Ensure that a destination callno of 0 will not match - for frames that do not start a dialog (new, lagrq, and ping). - (closes issue #12963) Reported by: russellb Patches: - chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) - ........ ................ - -2008-07-11 18:33 +0000 [r130168] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | - seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 - line Missed one. Formatting only. ........ - -2008-07-11 18:14 +0000 [r130130] Brett Bryant <bbryant@digium.com> - - * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c, - channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c, - codecs/codec_resample.c, codecs/codec_dahdi.c, - apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c, - main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c, - main/threadstorage.c, utils/astman.c, main/utils.c, - channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions - 130129 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | - bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines - Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue - #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 - uploaded by pabelanger (license 224) Tested by: seanbright - ........ - -2008-07-11 17:30 +0000 [r130127] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500 - (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) - | 9 lines Pass the devicestate from an underlying channel up - through the Agent channel. This should make the Agent always - report the correct device state, even when the underlying channel - is used for other purposes. (closes issue #12773) Reported by: - davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76 - (license 14) Tested by: davidw ........ ................ - -2008-07-11 16:18 +0000 [r129936-130045] Kevin P. Fleming <kpfleming@digium.com> - - * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES: - Merged revisions 130044 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 | - kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 - lines clean up a bunch more Zaptel-related references ........ - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, - configure, include/asterisk/autoconfig.h.in, configure.ac: Merged - revisions 130040 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul - 2008) | 12 lines Merged revisions 130039 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul - 2008) | 4 lines add support for a configuration parameter for - 'inband audio during RELEASE', which is currently mandatory in - libpri-1.4.4 but will become configurable in libpri-1.4.5 later - today (related to issue #13042) ........ ................ - - * /, main/astmm.c: Merged revisions 129968 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul - 2008) | 18 lines Merged revisions 129966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul - 2008) | 5 lines fix a flaw found while experimenting with - structure alignment and padding; low-fence checking would not - work properly on 64-bit platforms, because the compiler was - putting 4 bytes of padding between the fence field and the - allocation memory block added a very obvious runtime warning if - this condition reoccurs, so the developer who broke it can be - chastised into fixing it :-) ........ r129967 | kpfleming | - 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify - calculation ........ ................ - - * /, sounds/Makefile: Merged revisions 129916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul - 2008) | 10 lines Merged revisions 129907 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul - 2008) | 2 lines don't attempt to set user/group ownership of - extracted sound files (reported on asterisk-users) ........ - ................ - -2008-07-11 01:01 +0000 [r129865] Sean Bright <sean.bright@gmail.com> - - * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged - revisions 129864 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 | - seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 - line Fix some usages of snprintf, and clarify a couple variable - names. ........ - -2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500 - (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008) - | 8 lines Correctly deal with duplicate NEW frames (due to - retransmission). Also, fixup the destination call number matching - to be more strict and reliable. (closes issue #12963) Reported - by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by - jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........ - ................ - - * res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500 - (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008) - | 2 lines Oops ........ ................ - -2008-07-10 21:05 +0000 [r129739] Terry Wilson <twilson@digium.com> - - * Makefile, /: Merged revisions 129738 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 | - twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines - Move phoneprov config files to be installed with 'make samples' - so changes aren't inadvertently lost on a 'make install' ........ - -2008-07-10 19:14 +0000 [r129685] Brett Bryant <bbryant@digium.com> - - * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 | - bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines - Fixes a bug where the interface for a queue member gets reloaded - as the state_interface, if a state_interface was set, on reload - because the state_interface isn't stored in the ast_db. (closes - issue #13043) Reported by: jvandal Patches: app_queue.patch - uploaded by jvandal (license 413) ........ - -2008-07-10 18:20 +0000 [r129640-129647] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 | - seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 - line A couple more minor text changes ........ - - * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 | - seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 - line Remove extraneous \n. Pointed out by eliel on #asterisk-dev. - ........ - -2008-07-10 16:13 +0000 [r129570] Russell Bryant <russell@digium.com> - - * sample.call, /: Merged revisions 129569 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) - | 11 lines Merged revisions 129567 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) - | 3 lines Note that pbx_spool.so is the module used for call - files (inspired by a question in #asterisk) ........ - ................ - -2008-07-10 14:09 +0000 [r129504-129507] Sean Bright <sean.bright@gmail.com> - - * /, main/editline: Merged revisions 129503 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 | - seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 - lines Update svn:ignore ........ - -2008-07-09 19:41 +0000 [r129438] Mark Michelson <mmichelson@digium.com> - - * main/rtp.c, /: Merged revisions 129437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul - 2008) | 21 lines Merged revisions 129436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul - 2008) | 13 lines Fix a problem where inbound rfc2833 audio would - be sent to the core instead of being P2P bridged. When the core - regenerated the rfc2833 packet for the outbound leg, the SSRC - would be different than the RTP audio on the call leg causing - DTMF detection issues on the far end. (closes issue #12955) - Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by - tsearle (license 373) Tested by: tonyredstone ........ - ................ - -2008-07-09 16:01 +0000 [r129400] Matthew Fredrickson <creslin@digium.com> - - * main/pbx.c, /: Merged revisions 129399 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 | - mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add - Proceeding() application (#13025) ........ - -2008-07-09 13:46 +0000 [r129345] Sean Bright <sean.bright@gmail.com> - - * main/editline/makelist (removed), main/editline/makelist.in - (added), /, main/editline/configure, main/editline/Makefile.in, - main/editline/configure.in: Merged revisions 129344 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400 - (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul - 2008) | 4 lines Look for the system installed awk instead of - assuming it's at /usr/bin/awk. Pointed out by jmls via - #asterisk-dev. ........ ................ - -2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul - 2008) | 3 lines Fix compilation error when IMAP storage is - enabled ........ - -2008-07-08 21:04 +0000 [r129157] Brett Bryant <bbryant@digium.com> - - * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) - | 6 lines Fix a bug in SRV lookups where dnsmgr would discard - everything but the first SRV result from DNS before processing - weights and priorities and dns_parse_answer wouldn't report that - there were no records in DNS unless a failure occured. Also fixed - a bug where dnsmgr_refresh would report that a entry was being - changed when ast_gethostbyname had failed. ........ - -2008-07-08 20:31 +0000 [r129049-129153] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, /, channels/chan_sip.c, - include/asterisk/causes.h: Merged revisions 129152 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 - (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) - | 8 lines Cause SIP to return a 480 instead of a 404 when a sip - peer exists, but is not registered. (closes issue #12885) - Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded - by Corydon76 (license 14) Tested by: ibc ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500 - (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) - | 7 lines Timestamp decoding for video mini-frames is bogus, - because the timestamp only includes 15 bits, unlike voice frames, - which contain a 16-bit timestamp. (closes issue #13013) Reported - by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by - jpgrayson (license 492) ........ ................ - -2008-07-08 16:41 +0000 [r129041-129046] Brett Bryant <bbryant@digium.com> - - * main/rtp.c, main/channel.c, channels/chan_dahdi.c, - main/manager.c, formats/format_pcm.c, main/logger.c, - main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, - main/pbx.c, main/frame.c, /, channels/chan_sip.c, - apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c, - main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c, - channels/chan_iax2.c: Merged revisions 129045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | - bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines - Janitor project to convert sizeof to ARRAY_LEN macro. (closes - issue #13002) Reported by: caio1982 Patches: - janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ - - * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 | - bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line - Update transport= in sip so that the option is not broken from a - recent commit. ........ - - * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 | - bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line - Fix to sip_parse_host so that it passes the correct information - to sip_registry. ........ - -2008-07-08 14:18 +0000 [r129007] Russell Bryant <russell@digium.com> - - * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 | - russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines - Update app_fax for better compatibility with spandsp 0.0.5. Add a - call to t38_terminal_release, and make sure that the phase E - handler gets called with proper status. (closes issue #13020) - Reported by: dimas Patches: v1-appfax.patch uploaded by dimas - (license 88) ........ - -2008-07-08 10:06 +0000 [r128913-128952] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 - lines Merged revisions 128950 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 - lines Don't hangup the call if we can't resolve the Contact if - there's a proxy route set for the call. ---- This comment was - added a while ago and today it hit me badly. /* OEJ: Possible - issue that may need a check: If we have a proxy route between us - and the device, should we care about resolving the contact or - should we just send it? */ ........ ................ - - * /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 - lines Merged revisions 128912 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 - lines Fix issues where repeated messages where ignored, but - retransmitted reliably instead of unreliably. Reported by: johan - Patches: 12746.txt uploaded by oej (license 306) Tested by: johan - (issue #12746) ........ ................ - -2008-07-08 00:03 +0000 [r128855-128858] Tilghman Lesher <tlesher@digium.com> - - * /: Merged revisions 128857 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) - | 15 lines Merged revisions 128856 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) - | 7 lines Check for non-NULL before stripping characters. (closes - issue #12954) Reported by: bfsworks Patches: - 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) - Tested by: deti ........ ................ - - * apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500 - (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) - | 2 lines Stop using deprecated method, as requested by Kevin. - ........ ................ - -2008-07-07 22:44 +0000 [r128797] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r128796 | russell | 2008-07-07 17:42:30 -0500 - (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) - | 8 lines Fix handling of when a pvt disappears. Properly return - the pvt locked and don't hold the pvt lock while destroying the - ast_channel. (closes issue #13014) Reported by: jpgrayson - Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson - (license 492) ........ ................ - -2008-07-07 20:51 +0000 [r128739] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400 - (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul - 2008) | 9 lines Remove spurious trailing whitespace from log - messages and fix a spelling error in a log message. (closes issue - #13017) Reported by: jpgrayson Patches: - chan_iax2_space_after_newline.patch uploaded by jpgrayson - (license 492) chan_iax2_spelling.patch uploaded by jpgrayson - (license 492) ........ ................ - -2008-07-07 20:31 +0000 [r128601-128735] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul - 2008) | 3 lines Crap ........ - - * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul - 2008) | 7 lines If imapfolder=foo were set in voicemail.conf, - then when calling VoiceMailMain, app_voicemail would attempt to - play a file called vm-foo instead of playing vm-INBOX to play the - "new" sound file. This commit fixes that issue. This may fix one - of the problems reported in issue #12987 ........ - - * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500 - (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul - 2008) | 10 lines By using the iaxdynamicthreadcount to identify a - thread, it was possible for thread identifiers to be duplicated. - By using a globally-unique monotonically- increasing integer, - this is now avoided. (closes issue #13009) Reported by: jpgrayson - Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson - (license 492) ........ ................ - - * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged - revisions 128599 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 | - mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 - lines Update a few instances of "extensions reload" to "dialplan - reload" in the documentation. Patch provided by caio1982 (license - 22) ........ - -2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 128524 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 | - oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines - - Fixing issues with "sip show settings" - Adding IP address for - TCP and/or TLS too if auto-domain is enabled and binding to a - different IP address - Fixing documentation in sip.conf.sample - ........ - - * /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 | - oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines - - Remove unused variable "expiry" - Set global_outboundproxy.force - at reload. ........ - - * doc/realtimetext.txt (added), /: The following patch with - references to t140red removed, since it only exists in trunk. - Merged revisions 128417 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 | - oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines - Adding documentation on the T.140 support in Asterisk. This is a - function that we're the reference implementation on now. :-) - ........ - - * /: Merged revisions 128343 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 | - oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines - Removing the CLI dumpdb command (see asterisk-dev discussion and - decision) ........ - - * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 | - oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines - Adding doxygen comments to missing parts, moving some #define - ...trying to get my head around the thoughts behind the TCP/TLS - stuff and figure out what needs to be done to make it useful... - ........ - - * /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 | - oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines - Adding TCP and TLS to "sip show settings". TLS needs to have one - configuration per configured domain at some point. ........ - - * /: Blocking changes in trunk. - -2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson <oej@edvina.net> - - * /: Keep the "sip-user" structure in 1.6.0, while testing new - funky stuff in trunk. - - * /: Blocking the AGi changes from 1.6.0. Let's test them for a - while in trunk before a release. - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 128237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 | - oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines - Make TCP disabled by default (it's considered experimental) - ........ - - * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 - lines Reformatting the config sample ........ - -2008-07-05 15:19 +0000 [r128161] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/asterisk.ldap-schema, - contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 - Jul 2008) | 7 lines LDAP schema updates (closes issue #12860) - Reported by: flyn Patches: asterisk.ldif uploaded by suretec - (license 70) asterisk.schema uploaded by suretec (license 70) - ........ - -2008-07-05 03:40 +0000 [r128124-128127] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | - 1 line It would help if we actually parsed the ss7_explicitacm - option in the config file... ........ - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged - revisions 128122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 | - mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add - option to wait to be able to explicitly send ACM via the - Proceeding() application in the dialplan. Also minor - documentation update explaining how to setup multiple signalling - links within a linkset ........ - -2008-07-04 16:12 +0000 [r128028-128031] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged - revisions 128027 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) - | 16 lines Merged revisions 127973 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) - | 8 lines Fix the 'dialplan remove extension' logic, so that it - a) works with cidmatch, and b) completes contexts correctly when - the extension is ambiguous. (closes issue #12980) Reported by: - licedey Patches: 20080703__bug12980.diff.txt uploaded by - Corydon76 (license 14) Tested by: Corydon76 ........ - ................ - -2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged - revisions 127903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul - 2008) | 20 lines Merged revisions 127892,127895 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul - 2008) | 6 lines a couple of small Solaris-related fixes (closes - issue #11885) Reported by: snuffy, asgaroth ........ r127895 | - kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 - lines remove this, it has been moved to the main Makefile - ........ ................ - -2008-07-03 19:12 +0000 [r127830] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /, - channels/chan_sip.c, main/features.c, include/asterisk/cdr.h: - Merged revisions 127793 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | - 38 lines Merged revisions 127663 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | - 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) - Reported by: murf Tested by: murf, deeperror (closes issue - #12907) Reported by: falves11 Tested by: murf, falves11 (closes - issue #11849) Reported by: greyvoip As to 11849, I think these - changes fix the core problems brought up in that bug, but perhaps - not the more global problems created by the limitations of CDR's - themselves not being oriented around transfers. Reopen if necc, - but bug reports are not the best medium for enhancement - discussions. We need to start a second-generation CDR - standardization effort to cover transfers. (closes issue #11093) - Reported by: rossbeer Tested by: greyvoip, murf ........ - ................ - -2008-07-03 16:50 +0000 [r127790-127792] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 | - oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make - sure we stop session timers as soon as we start hanging up an - active call. May fix issue 12919. ........ - - * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 | - oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines - Revert some logic for session timers. We do send in-dialog - requests that should not have session-timer require headers, like - MESSAGE and REFER. So in the future, only add them on requests - and responses that are related to INVITEs and re-INVITEs. - ........ - -2008-07-03 16:24 +0000 [r127778] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: Merged revisions 127767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | - kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 - lines some minor fixes found while working on issue #12911 (and - block the rev from 1.4 since the equivalent is already here) - ........ - -2008-07-02 21:10 +0000 [r127567] Mark Michelson <mmichelson@digium.com> - - * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul - 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of - in-line sizeof() and division. ........ - -2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500 - (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul - 2008) | 3 lines Fix thread-safety of some of the - pbx_builtin_getvar_helper calls ........ ................ - -2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher <tlesher@digium.com> - - * /, main/acl.c: Merged revisions 127466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | - tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines - Solaris fix (closes issue #12949) Reported by: snuffy Patches: - bug_12949.diff uploaded by snuffy (license 35) ........ - -2008-07-02 14:30 +0000 [r127396-127399] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 | - seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 - line Fix a bug I noticed while doing the previous merge ........ - - * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure, - include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt: - Merged revisions 126226,126513 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | - seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 - lines Merge in changes from my cdr-tds-conversion branch. This - changes the internal implementation from using the volatile - libtds, to using the db-lib front end. The unintended side effect - of this is that we support (at least) versions 0.62 through 0.82 - of the FreeTDS distribution without any #ifdef ugliness. (closes - issue #12844) Reported by: jcollie ........ r126513 | seanbright - | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a - few more strings to char *, so that we can compile cleanly - against FreeTDS 0.60. Update the docs to reflect that we can now - compile and run against all modern releases of FreeTDS (0.60 - through 0.82) ........ - - * /: Unblock some revisions so I can merge the cdr_tds changes from - trunk - -2008-07-02 12:09 +0000 [r127364] Russell Bryant <russell@digium.com> - - * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) - | 13 lines Add a locking section to the coding guidelines - document. This section covers some locking fundamentals, as well - as some information on locking as it is used in Asterisk. It - describes some of the ways that are used and could be used to - achieve deadlock avoidance. It also demonstrates the unfortunate - conclusion that with the use of recursive locks, none of the - constructs in use today are failsafe from deadlocks. Finally, it - makes some recommendations for new code being written. As proper - locking strategies is a complex subject, this section still has - room for expansion and improvement. This is a result of - collaboration between Luigi Rizzo and myself on the asterisk-dev - mailing list. ........ - -2008-07-02 02:49 +0000 [r127298] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 | - tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 - lines Change the global timer B to be dependent on the value of - the T1 timer, as recommended in RFC 3261, instead of being - hardcoded to 32 seconds. This is important for LANs, as it allows - autocongestion to occur much more quickly, if desired by the - local PBX administrator. It also corrects a bug: if the T1 timer - was increased beyond 500ms, then timer B would have been set at a - much lower value than recommended. (closes issue #12544) Reported - by: kactus Patches: 20080616__bug12544.diff.txt uploaded by - Corydon76 (license 14) Tested by: kactus ........ - -2008-07-01 23:39 +0000 [r127246] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500 - (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul - 2008) | 5 lines Add error message to failed open(2) calls inside - the copy() function of app_voicemail. This idea came as part of - my work in helping to resolve issue #12764. ........ - ................ - -2008-07-01 21:19 +0000 [r127163] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 127154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 | - bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines - Add a configuration option so the global outboundproxy can use - tcptls without it being defined by each sip user. ........ - -2008-07-01 21:16 +0000 [r127156-127158] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 127157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 | - mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8 - lines Place the delay in __ast_answer prior to the - channel-specific answer callback. This change differs from commit - 127113 in that now the channel is not set to AST_STATE_UP until - after the answer callback. (closes issue #12924) Reported by: - snyfer ........ - - * main/channel.c, /: Merging Revision 127113 from trunk - -2008-07-01 20:52 +0000 [r127153] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 127152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 | - qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines - Fix a typo that caused this asterisk.conf to not get correctly - generated. (closes issue #12966) Reported by: ibc Patches: - 12966.patch uploaded by bkruse (license 132) ........ - -2008-07-01 20:29 +0000 [r127085-127149] Tilghman Lesher <tlesher@digium.com> - - * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions - 127143 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008) - | 10 lines Merged revisions 127133 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) - | 2 lines Disable the old, slow search for matching callno in - chan_iax2 (but allow it to be reenabled for debugging) ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500 - (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) - | 8 lines Change around how we schedule pings and lagrqs, and fix - a reason why the jobs were not getting properly cancelled. - (closes issue #12903) Reported by: stevedavies Patches: - 20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: stevedavies ........ ................ - -2008-07-01 16:53 +0000 [r127001] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500 - (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) - | 2 lines Suppress annoying warning by finding the remaining - cases where the callno is not in the hash. ........ - ................ - -2008-07-01 15:05 +0000 [r126756-126904] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15 - lines Merged revisions 126902 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 - lines Use domain part of SIP uri in register= configuration as - fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch - uploaded by one47 (license 23) (closes issue #12474) ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16 - lines Merged revisions 126899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 - lines Handle escaped URI's in call pickups. Patch by oej and - IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch - uploaded by IgorG (license 20) Tested by: IgorG, oej (closes - issue #12299) ........ ................ - - * /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, - 01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 - lines Clear up documentation on "domain=" setting in sip.conf - Reported by: davidw (closes issue #12413) ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 - lines Merged revisions 126789 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 - lines Report 200 OK to all in-dialog OPTIONs requests (to confirm - that the dialog exist). Don't bother checking the request URI. - (closes issue #11264) Reported by: ibc ........ ................ - - * /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15 - lines Merged revisions 126735 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 - lines Fix bad XML for hold notification. Reported by: gowen72 - Patches: hold.patch uploaded by gowen72 (license 432) (closes - issue #12942) ........ ................ - -2008-06-30 22:34 +0000 [r126676] Jeff Peeler <jpeeler@digium.com> - - * configs/zapata.conf.sample (removed), - configs/chan_dahdi.conf.sample (added), /: Merged revisions - 126675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 | - jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line - rename zapata.conf.sample to chan_dahdi.conf.sample ........ - -2008-06-30 20:32 +0000 [r126638] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) | - 1 line Add support to see MTP2 down events when the link layer - drops in SS7 ........ - -2008-06-30 16:09 +0000 [r126575] Russell Bryant <russell@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r126574 | russell | 2008-06-30 11:07:25 -0500 - (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) - | 10 lines Fix a typo in the non-DEBUG_THREADS version of the - recently added DEADLOCK_AVOIDANCE() macro. This caused the lock - to not actually be released, and as a result, not avoid deadlocks - at all. This resolves the issues reported in the last while about - Asterisk locking up all over the place (and most commonly, in - chan_iax2). (closes issue #12927) (closes issue #12940) (closes - issue #12925) (potentially closes others ...) ........ - ................ - -2008-06-30 13:07 +0000 [r126518] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | - 20 lines The following patch with some changes for trunk... - Merged revisions 126516 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | - 10 lines Send all responses to an INVITE reliably, so that we - retransmit if we don't get an ACK and also fail if we don't get - the very same precious ACK. Based on patch by tsearle, with my - own additions. (closes issue #12951) Reported by: tsearle - Patches: busy_retransmit.patch uploaded by tsearle (license 373) - ........ ................ - -2008-06-29 17:02 +0000 [r126362-126364] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_zapbarge.c (removed): finish converting this module - - * pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c, - pbx/Makefile: Merged revisions 126356 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 | - kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9 - lines various minor fixes created while i worked on getting - *every* Asterisk module to build on laptop in dev mode: remove - weird pre-setting of LUA paths; they are not necessary; also use - the proper path for including the files in pbx_lua.c make the - compiler shut up about some ignored function results in - pbx_gtkconsole; this module is badly coded anyway ........ - - * apps/app_dahdibarge.c (added): don't know how this got missed in - the DAHDI conversion of this branch - -2008-06-29 13:20 +0000 [r126227-126322] Sean Bright <sean.bright@gmail.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 | - seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6 - lines Quote column names when inserting CDRs into postgres to - avoid conflicts with reserved words. (closes issue #12947) - Reported by: panolex ........ - -2008-06-28 15:58 +0000 [r126155-126188] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: update this branch to use the trunk goodness version - of menuselect - -2008-06-27 22:43 +0000 [r126058-126112] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008) - | 12 lines Merged revisions 126056 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) - | 4 lines When we get a 408 Timeout, don't stop trying to - re-register. (closes issue #12863) Reported by: ricvil ........ - ................ - -2008-06-27 21:00 +0000 [r126023] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Port revisions 124661 and 123650 from trunk to - 1.6.0 Thanks to Atis Lezdins for pointing this out on the - asterisk-dev mailing list - -2008-06-27 19:20 +0000 [r125994] Russell Bryant <russell@digium.com> - - * /, doc/siptls.txt: Merged revisions 125988 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 | - russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines - Fix a typo. Someone on IRC copied this literally and then - wondered why it wasn't working. :) ........ - -2008-06-27 19:06 +0000 [r125981-125985] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) | - 1 line Revert this part of the fix. We'll fix it in libss7 - ........ - - * channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) | - 1 line Obviously somebody didn't compile with libss7 support when - doing the DAHDI conversion. ........ - - * channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) | - 1 line Add support for new commands to block/unblock all CICs on - a linkset ........ - -2008-06-27 17:36 +0000 [r125948] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 | - bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line - Small error in the function that converts peer transports to a - string. ........ - -2008-06-27 16:29 +0000 [r125892] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 125891 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 | - bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines - Change the way that the transport option works for sip users. - transport will now take multiple arguments, the first one listed - will be the one used for new dialogs, and the rest listed will be - acceptable ways for that peer to contact us. This fixes a minor - bug where, because SIP TCP/UDP run on the same port, could cause - a TCP peer to be saved in the ast_db. There will also be warnings - when a transport is changed for an unexpected reason. (issue - #12799) ........ - -2008-06-27 16:19 +0000 [r125859-125863] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 125855 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 | - mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5 - lines Ensure the thread-safety of the monexec variable in - app_queue. Thanks to Russell for pointing out the problem - ........ - -2008-06-27 16:01 +0000 [r125854] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 125853 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008) - | 3 lines Revert half of the fix, as this part may have been - unnecessary (related to issue #12914) Requested here: - http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html - ........ - -2008-06-27 14:57 +0000 [r125800-125852] Mark Michelson <mmichelson@digium.com> - - * main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure - to only include dahdi/user.h if we have installed DAHDI. - - * channels/chan_iax2.c: I accidentally committed a change to - chan_iax2.c in addition to a change to app_queue.c. Reverting the - change to chan_iax2.c, even though it may turn out that this - change is necessary. - - * utils/Makefile, /: Merged revisions 125799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 | - mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3 - lines Remove an unneeded target from the Makefile ........ - -2008-06-27 14:09 +0000 [r125742-125797] Tilghman Lesher <tlesher@digium.com> - - * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500 - (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) - | 2 lines In this debugging function, copy to a buffer instead of - using potentially unsafe pointers. ........ ................ - - * channels/chan_local.c, /: Merged revisions 125741 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500 - (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) - | 7 lines Add proper deadlock avoidance. (closes issue #12914) - Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded - by Corydon76 (license 14) Tested by: ozan ........ - ................ - -2008-06-27 07:41 +0000 [r125704] Philippe Sultan <philippe.sultan@gmail.com> - - * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions - 125703 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 | - phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line - Fix a compile time error that occurs if OpenSSL is not installed. - Reported by Noel Morais on the users mailing list ........ - -2008-06-27 01:09 +0000 [r125648-125684] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined - in 1.6.0 - - * /, apps/app_queue.c: Merged revisions 125666 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 | - mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3 - lines Make this compile with dev-mode on ........ - - * /, apps/app_queue.c: Merged revisions 125649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 | - mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 - lines The monitor-join option for queues was deprecated in favor - of using MixMonitor to mix audio. However, it was pointed out to - me that because of this, the command set for the MONITOR_EXEC - variable is ignored as well. This means that people can't do - their own custom mixing commands at the end of recordings in - order to make, for instance, stereo recordings of calls. With - this patch, app_queue will set the "joinfiles" variable for the - channel's monitor if MONITOR_EXEC is not zero-length. This means - that for normal audio mixing, MixMonitor is still the preferred - choice, but we allow custom mixing to be done with the two - Monitor streams if desired. (closes issue #12923) Reported by: - snyfer ........ - -2008-06-26 23:06 +0000 [r125592] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 125591 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 | - mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3 - lines Fix a really stupid mistake ........ - -2008-06-26 23:05 +0000 [r125590] Jason Parker <jparker@digium.com> - - * /, main/utils.c: Merged revisions 125589 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) | - 9 lines Merged revisions 125587 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | - 1 line Make sure to unlock the lock_info lock (huh?). Possible - deadlock? ........ ................ - -2008-06-26 23:04 +0000 [r125588] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 125586 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun - 2008) | 19 lines Merged revisions 125585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun - 2008) | 11 lines Add the interface of a queue member to the - output of the "queue show" command so that it can easily be - associated with a queue member's name. This helps so that the - appropriate queue member can be removed or paused since the - interface is required, not the member's name. (closes issue - #12783) Reported by: davevg Patches: app_queue.diff uploaded by - davevg (license 209) with small mod from me ........ - ................ - -2008-06-26 22:50 +0000 [r125584] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008) - | 2 lines Don't hang if the command is blank ........ - -2008-06-26 22:06 +0000 [r125478-125532] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 125477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun - 2008) | 19 lines Merged revisions 125476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun - 2008) | 11 lines Prior to this patch, the "queue show" command - used cached information for realtime queues instead of giving - up-to-date info. Now realtime is queried for the latest and - greatest in queue info. (closes issue #12858) Reported by: bcnit - Patches: queue_show.patch uploaded by putnopvut (license 60) - ........ ................ - -2008-06-26 17:07 +0000 [r125388] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12 - lines Merged revisions 125384 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 - lines Add support for peer realm based auth (a few missing lines, - the rest is well documented but never worked) ........ - ................ - -2008-06-26 15:52 +0000 [r125280-125334] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500 - (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun - 2008) | 5 lines ensure that (whenever possible) if we generate a - log message because an ioctl() call to DAHDI/Zaptel failed, that - we include the reason it failed by including the stringified - error number (issue AST-80) ........ ................ - - * /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun - 2008) | 2 lines fix compile failure found by buildbot (go, - buildbot!) ........ - -2008-06-26 11:08 +0000 [r125192-125278] Tilghman Lesher <tlesher@digium.com> - - * main/rtp.c, /: Merged revisions 125277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) - | 15 lines Merged revisions 125276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) - | 7 lines Check for rtcp structure before trying to delete - schedule. (closes issue #12872) Reported by: destiny6628 Patches: - 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) - Tested by: destiny6628 ........ ................ - - * configs/agents.conf.sample, /: Merged revisions 125223 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500 - (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) - | 4 lines Document ackcall=always. (closes issue #12852) Reported - by: davidw ........ ................ - - * configs/http.conf.sample, /: Merged revisions 125191 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) - | 6 lines Update sample configuration to match what are now the - defaults for the prefix. (closes issue #12838, related to issue - #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded - by pabelanger (license 224) ........ - -2008-06-25 23:20 +0000 [r125146] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, channels/chan_dahdi.c, apps/app_flash.c, - configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c, - /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c, - apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h - (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged - revisions 125138 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun - 2008) | 18 lines Merged revisions 125132 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun - 2008) | 10 lines allow tonezone to live in a different place than - DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate - packages and can be installed in different places don't include - tonezone.h in dahdi_compat.h, because only a couple of modules - need it get app_rpt building again after the DAHDI changes - (closes issue #12911) Reported by: tzafrir ........ - ................ - -2008-06-25 01:13 +0000 [r124964-124967] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged - revisions 124966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) - | 15 lines Merged revisions 124965 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) - | 7 lines Pvt deadlock causes some channels to get stuck in - Reserved status. (closes issue #12621) Reported by: - fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by - Corydon76 (license 14) Tested by: fabianoheringer ........ - ................ - - * apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500 - (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) - | 8 lines Occasionally control characters find their way into - CallerID. These need to be stripped prior to placing CallerID in - the headers of an email. (closes issue #12759) Reported by: RobH - Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76 - (license 14) Tested by: RobH ........ ................ - -2008-06-24 17:52 +0000 [r124871-124873] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 124872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 | - phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines - Subscribe to buddy's presence only if we really need to. That is, - if the corresponding roster item has a subscription value set to - "none" or "from". Make the code more readable. ........ - - * /, res/res_jabber.c: Merged revisions 124870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 | - phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line - Code simplification ........ - -2008-06-23 15:44 +0000 [r124708] Dwayne M. Hubbard <dhubbard@digium.com> - - * /: blocked revision 124707, taskprocessors are not in 1.6.0 - -2008-06-22 03:18 +0000 [r124542] Steve Murphy <murf@digium.com> - - * apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) | - 17 lines Merged revisions 124540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 - lines (closes issue #12910) Reported by: chris-mac Sorry, my - testing did not contain the simple case of forkCDR(v), I am much - embarrassed to admit. If I had, I would have more solidly - initialized the opts element for varset. ........ - ................ - -2008-06-21 12:54 +0000 [r124397-124506] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008) - | 4 lines Reduce warning to debug, otherwise we flood the log - when we (legitimately) can't find a record. (Closes issue #12908) - ........ - - * apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008) - | 14 lines Merged revisions 124450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) - | 6 lines usleep with a value over 1,000,000 is nonportable. - Changing to use sleep() instead. (closes issue #12814) Reported - by: pputman Patches: app_rtp_sleep.patch uploaded by pputman - (license 81) ........ ................ - - * /, main/app.c: Merged revisions 124396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008) - | 11 lines Merged revisions 124395 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) - | 3 lines If the last character in a string to be parsed is the - delimiter, then we should count that final empty string as an - additional argument. ........ ................ - -2008-06-20 21:48 +0000 [r124394] Jeff Gehlbach <jeffg@opennms.org> - - * doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions - 124392-124393 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) | - 9 lines Merged revisions 124372 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | - 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to - placate smilint - bug 12905 ........ ................ r124393 | - jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines - (Missed committing . on previous commit.....) Merged revisions - 124372 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | - 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to - placate smilint - bug 12905 ........ ................ - ................ - -2008-06-20 20:18 +0000 [r124317] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 124316 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500 - (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) - | 8 lines When using a Local channel, started by a call file, - with a destination of an AGI script, the AGI script does not - always get notified of a hangup if the underlying channel hangs - up early. (closes issue #11833) Reported by: IgorG Patches: - local_hangup-v1.diff uploaded by IgorG (license 20) ........ - ................ - -2008-06-20 16:31 +0000 [r124244-124279] Mark Michelson <mmichelson@digium.com> - - * main/ast_expr2.fl, include/asterisk/doxyref.h, /, - main/ast_expr2f.c: Merged revisions 124278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 | - mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 - lines Change references to doc/channelvariables.txt to - doc/tex/channelvariables.tex. This issue came up on the - asterisk-dev mailing list. ........ - - * /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 | - mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9 - lines Add a missing "ChannelType" header to one of the - "PeerStatus" manager events in chan_sip (closes issue #12904) - Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel - (license 64) ........ - -2008-06-19 23:02 +0000 [r124184] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500 - (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) - | 7 lines It's possible for a hangup to be received, even just - after the initial cid spill. (closes issue #12453) Reported by: - Alex728 Patches: 20080604__bug12453.diff.txt uploaded by - Corydon76 (license 14) ........ ................ - -2008-06-19 20:32 +0000 [r124124] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500 - (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun - 2008) | 8 lines Fix IMAP forwarding so that messages are sent to - the proper mailbox. (closes issue #12897) Reported by: jaroth - Patches: destination_forward.patch uploaded by jaroth (license - 50) ........ ................ - -2008-06-19 19:49 +0000 [r124065] Brett Bryant <bbryant@digium.com> - - * /, main/utils.c: Merged revisions 124064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 | - bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines - Add errors that report any locks held by threads when they are - being closed. ........ - -2008-06-19 18:57 +0000 [r124026] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 | - bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines - Fix bug in sip registration that sets the default port to 5060 - for tls. ........ - -2008-06-19 17:58 +0000 [r123871-123989] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008) - | 6 lines Don't change pointers that need to be later passed back - for deallocation. (closes issue #12572) Reported by: flyn - Patches: 20080613__bug12572.diff.txt uploaded by Corydon76 - (license 14) ........ - - * main/channel.c, /: Merged revisions 123931 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008) - | 13 lines Merged revisions 123930 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) - | 5 lines Change informative messages to use the _multiple - variant when multiple formats are possible. (Closes issue #12848) - Reported by klaus3000 ........ ................ - - * /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500 - (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008) - | 5 lines Only process 40 arguments (20 files) at once with - xargs, because some older shells may force xargs to separate on - an odd boundary. (Closes issue #12883) Reported by Nik Soggia - ........ ................ - - * /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500 - (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) - | 4 lines Correct description of notifyringing option. (Closes - issue #12890) Reported by gminet ........ ................ - - * main/asterisk.c, /: Merged revisions 123870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008) - | 14 lines Merged revisions 123869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) - | 6 lines The RDTSC instruction was introduced on the Pentium - line of microprocessors, and is not compatible with certain 586 - clones, like Cyrix. Hence, asking for i386 compatibility was - always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes - issue #12886) Reported by tecnoxarxa ........ ................ - -2008-06-18 22:18 +0000 [r123718-123772] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged - revisions 123770 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008) - | 16 lines Merged revisions 123769 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) - | 8 lines Add support for saying numbers in Hebrew. (closes issue - #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 - uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods - uploaded by greenfieldtech (with signficant changes to the - spreadsheet by me) ........ ................ - - * pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008) - | 15 lines Merged revisions 123710 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) - | 7 lines Set the variables top-down, so that if a script sets a - variable more than once, the last one will take precedence. - (closes issue #12673) Reported by: phber Patches: - 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) - ........ ................ - -2008-06-18 20:08 +0000 [r123693] Brett Bryant <bbryant@digium.com> - - * main/tcptls.c, /: Merged revisions 123692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 | - bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines - Fix a crash in tcp and tls connections related to reference - counts. ........ - -2008-06-18 15:09 +0000 [r123651-123653] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 123652 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 | - mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7 - lines A portion of the code which handled the 'c' queue option - had been removed. No telling when it happened. Anyway, it's back - in now and works properly. (Based on issue reported on mailing - list) ........ - -2008-06-18 12:34 +0000 [r123646-123647] Russell Bryant <russell@digium.com> - - * apps/app_fax.c: don't use trunk only API for frame data (closes - issue #12881) - - * apps/app_fax.c (added): re-add app_fax ... it got accidentally - removed (closes issue #12881) - -2008-06-17 21:57 +0000 [r123547] Brett Bryant <bbryant@digium.com> - - * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, - main/http.c, include/asterisk/tcptls.h: Merged revisions 123546 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 - Jun 2008) | 5 lines Updates all usages of - ast_tcptls_session_instance to be managed by reference counts so - that they only get destroyed when all threads are done using - them, and memory does not get free'd causing strange issues with - SIP. This code was originally written by russellb in the - team/group/issue_11972/ branch. ........ - -2008-06-17 21:34 +0000 [r123487-123542] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun - 2008) | 12 lines Merged revisions 123485 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun - 2008) | 4 lines Make chan_sip build under dev mode with compilers - >= GCC 4.2 Thanks to jpeeler for alerting me of this ........ - ................ - -2008-06-17 20:23 +0000 [r123473] Steve Murphy <murf@digium.com> - - * /: block 123448 from trunk; it doesn't apply here. - -2008-06-17 19:01 +0000 [r123394] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500 - (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) - | 3 lines Fix 3 more places where failure to lock the structure - could cause the wrong lock to be unlocked. (Closes issue #12795) - ........ ................ - -2008-06-17 18:28 +0000 [r123382-123387] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008) - | 1 line Fix some (more) variables that were forgotten to be - renamed, related to 117658 ........ - -2008-06-17 18:10 +0000 [r123335] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun - 2008) | 19 lines Merged revisions 123333 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun - 2008) | 11 lines Cisco BTS sends SIP responses with a tab between - the Cseq number and SIP request method in the Cseq: header. - Asterisk did not handle this properly, but with this patch, all - is well. (closes issue #12834) Reported by: tobias_e Patches: - 12834.patch uploaded by putnopvut (license 60) Tested by: - tobias_e ........ ................ - -2008-06-17 18:08 +0000 [r123332] Jeff Peeler <jpeeler@digium.com> - - * doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile, - doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt, - contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added), - configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c, - main/asterisk.c, configs/users.conf.sample, doc/ss7.txt, - apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt, - doc/tex/queues-with-callback-members.tex, - include/asterisk/dahdi.h (added), configs/extensions.ael.sample, - res/res_musiconhold.c, configs/meetme.conf.sample, - codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel, - cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex, - doc/tex/sla.tex, include/asterisk/dsp.h, - doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c, - contrib/scripts/autosupport, doc/manager_1_1.txt, - channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex, - doc/tex/channelvariables.tex, apps/app_getcpeid.c, - doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample, - doc/tex/security.tex, include/asterisk/zapata.h (removed), - doc/tex/privacy.tex, build_tools/menuselect-deps.in, - apps/app_flash.c, main/file.c, doc/osp.txt, - contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in, - configs/extensions.conf.sample, doc/asterisk.sgml, README, - contrib/init.d/rc.mandrake.asterisk, /, - include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added), - apps/app_chanisavail.c, channels/chan_iax2.c, - configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c - (added), include/asterisk/doxyref.h, configure, - doc/tex/backtrace.tex, apps/app_zapscan.c (removed), - doc/tex/app-sms.tex, apps/app_zapras.c (removed), - configs/extensions.lua.sample, include/asterisk/options.h, - contrib/init.d/rc.suse.asterisk, apps/app_dial.c, - apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed), - apps/app_dahdiras.c (added), configure.ac, - configs/queues.conf.sample, include/asterisk/channel.h: Goodbye - Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. - Configuration file and dialplan backwards compatability has been - put in place where appropiate. Release announcement to follow. - -2008-06-17 15:58 +0000 [r123276] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 123275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun - 2008) | 20 lines Merged revisions 123274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun - 2008) | 12 lines davidw pointed out that the holdtime calculation - used by app_queue does not use "boxcar" filtering as the comments - say. The term "boxcar" means that the number of samples used to - calculate stays constant, with new samples replacing the oldest - ones. The queue holdtime calculation uses all holdtime samples - collected since the queue was loaded, so the comment has been - changed to be accurate. (closes issue #12781) Reported by: davidw - ........ ................ - -2008-06-17 15:52 +0000 [r123273] Russell Bryant <russell@digium.com> - - * main/astobj2.c, /: Merged revisions 123272 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) - | 12 lines Merged revisions 123271 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) - | 4 lines Fix a memory leak in astobj2 that was pointed out by - seanbright. When a container got destroyed, the underlying bucket - list entry for each object that was in the container at that time - did not get free'd. ........ ................ - -2008-06-16 21:20 +0000 [r123178] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Fix some variables that were forgotten to be - renamed, related to 117658. Couldn't merge from trunk since the - chan_dahdi transition has not occurred here yet - -2008-06-16 21:19 +0000 [r123173] Steve Murphy <murf@digium.com> - - * apps/app_stack.c, apps/app_dial.c, main/pbx.c, /, - main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged - revisions 123165 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 | - murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines - (closes issue #12689) Reported by: ys Many thanks to ys for doing - the research on this problem. I didn't think it would be best to - unlock the contexts and then relock them after the - remove_extension2() call, so I added an extra arg to - remove_extension2() and set it appropriately in each call. There - were not that many. I considered forcing the code to lock the - contexts before the call to remove_extension2(), but that would - require a slightly greater degree of changes, especially since - the find_context_locked is local to pbx.c I did a simple sanity - test to make sure the code doesn't mess things up in general. - ........ - -2008-06-16 20:03 +0000 [r123112-123116] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_mgcp.c, /, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_iax2.c: Merged revisions 123114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008) - | 10 lines Merged revisions 123113 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) - | 2 lines Port "hasvoicemail" change from SIP to other channel - drivers ........ ................ - - * /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) - | 16 lines Merged revisions 123110 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) - | 8 lines People expect that if "hasvoicemail" is set in - users.conf, even if "mailbox" isn't set, that SIP will detect a - mailbox. (closes issue #12855) Reported by: PLL Patches: - 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: PLL ........ ................ - -2008-06-16 17:29 +0000 [r123075] Chris Tooley <chris@tooley.com> - - * apps/app_externalivr.c: Fixes and closes bug number 12804 - -2008-06-16 12:32 +0000 [r122871-122921] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | - 14 lines Merged revisions 122919 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 - lines Only compare the first 15 characters so that even if the - charset is specified we still accept it as SDP. (closes issue - #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff - uploaded by lanzaandrea (license 496) ........ ................ - - * /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) | - 14 lines Merged revisions 122869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 - lines Don't send a BYE on a dialog that is already gone during a - REFER. (closes issue #12865) Reported by: flefoll Patches: - chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll - (license 244) ........ ................ - -2008-06-13 21:47 +0000 [r122715] Mark Michelson <mmichelson@digium.com> - - * main/autoservice.c, /: Merged revisions 122714 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun - 2008) | 17 lines Merged revisions 122713 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun - 2008) | 9 lines Short circuit the loop in autoservice_run if - there are no channels to poll. If we continued, then the result - would be calling poll() with a NULL pollfd array. While this is - fine with POSIX's poll(2) system call, those who use Asterisk's - internal poll mechanism (Darwin systems) would have a failed - assertion occur when poll is called. (related to issue #10342) - ........ ................ - -2008-06-13 14:15 +0000 [r122558] Tilghman Lesher <tlesher@digium.com> - - * main/dial.c, /: Merged revisions 122557 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 | - tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines - Convert one more delimiter to use comma. (closes issue #12850) - Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded - by Corydon76 (license 14) Tested by: bcnit ........ - -2008-06-13 00:18 +0000 [r122467] Jeff Peeler <jpeeler@digium.com> - - * apps/app_parkandannounce.c, /, main/features.c: Merged revisions - 122433 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 | - jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines - (closes issue 0012193) Reported by: davidw Patch by: Corydon76, - modified by me to work properly with ParkAndAnnounce app ........ - -2008-06-12 18:54 +0000 [r122313] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 122312 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun - 2008) | 17 lines Merged revisions 122311 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun - 2008) | 9 lines Properly play a holdtime message if the - announce-holdtime option is set to "once." (closes issue #12842) - Reported by: ramonpeek Patches: patch001.diff uploaded by - ramonpeek (license 266) ........ ................ - -2008-06-12 18:24 +0000 [r122242-122266] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r122262 | russell | 2008-06-12 13:23:54 -0500 - (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) - | 3 lines Fix some race conditions that cause ast_assert() to - report that chan_iax2 tried to remove an entry that wasn't in the - scheduler ........ ................ - -2008-06-12 15:27 +0000 [r122132-122180] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) - | 16 lines Merged revisions 122137 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) - | 8 lines Flipflop the sections for two options, since the - section for 'X' (exit context) may otherwise absorb keypresses - meant for 's' (admin/user menu). (closes issue #12836) Reported - by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by - Corydon76 (license 14) Tested by: blitzrage ........ - ................ - - * main/channel.c, /: Merged revisions 122131 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008) - | 12 lines Merged revisions 122130 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) - | 4 lines Occasionally, the alertpipe loses its nonblocking - status, so detect and correct that situation before it causes a - deadlock. (Reported and tested by ctooley via #asterisk-dev) - ........ ................ - -2008-06-12 15:01 +0000 [r122126-122129] Steve Murphy <murf@digium.com> - - * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions - 122128 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 - lines Merged revisions 122127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 - line Arkadia tried to warn me, but the code added to - ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot - it until I was resolving conflicts in trunk. Ugh. Redundant code - removed. It wasn't harmful. Just dumb. ........ ................ - - * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c, - include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, - 12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | - 37 lines (closes issue #10668) Reported by: arkadia Tested by: - murf, arkadia Options added to forkCDR() app and the CDR() func - to remove some roadblocks for CDR applications. The "show - application ForkCDR" output was upgraded to more fully explain - the inner workings of forkCDR. The A option was added to forkCDR - to force the CDR system to NOT change the disposition on the - original CDR, after the fork. This involves ast_cdr_answer, - _busy, _failed, and so on. The T option was added to forkCDR to - force obedience of the cdr LOCKED flag in the ast_cdr_end, all - the disposition changing funcs (ast_cdr_answer, etc), and in the - ast_cdr_setvar func. The CHANGES file was updated to explain ALL - the new options added to satisfy this bug report (and some - requests made verbally and via email, irc, etc, over the past - months/year) The 's' option was added to the CDR() func, to force - it to skip LOCKED cdr's in the chain. Again, the new options - should be totally transparent to existing apps! Current behavior - of CDR, forkCDR, and the rest of the CDR system should not change - one little bit. Until you add the new options, at least! ........ - ................ - -2008-06-11 18:57 +0000 [r121915] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 | - mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix - pseudo channel allocation errors on startup when using SS7 - ........ - -2008-06-11 18:20 +0000 [r121872] Tilghman Lesher <tlesher@digium.com> - - * main/sched.c, main/channel.c, /, channels/chan_agent.c, - main/abstract_jb.c: Merged revisions 121867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008) - | 11 lines Merged revisions 121861 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) - | 3 lines Make calls to ast_assert() actually test something, so - that the error message printed is not nonsensical (reported by - mvanbaak via #asterisk-bugs). ........ ................ - -2008-06-11 17:59 +0000 [r121858] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 | - mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line - Make sure we hangup any calls we have and NULL out the ss7call - value when we get a reset circuit message. Fixes crash bug - ........ - -2008-06-11 17:45 +0000 [r121856] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt, - include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 | - tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines - Expand CDR uniqueid field to 150 chars, to account for maximum - systemname. (Closes issue #12831) ........ - -2008-06-11 16:13 +0000 [r121806] Jeff Peeler <jpeeler@digium.com> - - * /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008) - | 9 lines Merged revisions 121804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) - | 1 line add instructions for logging gdb output via set logging - on ........ ................ - -2008-06-10 18:36 +0000 [r121598] Sean Bright <sean.bright@gmail.com> - - * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400 - (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun - 2008) | 6 lines Fixes a problem with some buggy versions of GNU - awk (3.1.3) not liking carriage returns in scripts. (closes issue - #12749) Reported by: alinux Tested by: Laureano (on - #asterisk-dev), juggie ........ ................ - -2008-06-10 12:55 +0000 [r121445] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 121444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) | - 12 lines Merged revisions 121442 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 - lines Update BRIDGEPEER variable before we do a generic bridge in - case we just broke out of a native bridge and fell through to - generic. (closes issue #12815) Reported by: ramonpeek ........ - ................ - -2008-06-10 00:53 +0000 [r121404-121408] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008) - | 2 lines Bump up the debug level of a couple of messages - ........ - -2008-06-09 16:37 +0000 [r121283] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 121282 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) - | 18 lines Merged revisions 121280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) - | 10 lines Do not attempt to do emulation if an END digit is - received and the length is less than the defined minimum digit - length, and the other end only wants END digits (SIP INFO, for - example). (closes issue #12778) Reported by: tsearle Patches: - 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle - ........ ................ - -2008-06-09 16:36 +0000 [r121281] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 121279 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 | - tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines - Implement FINDLABEL matching for the new extension matching - engine. (closes issue #12800) Reported by: chris-mac Patches: - 20080608__bug12800.diff.txt uploaded by Corydon76 (license 14) - ........ - -2008-06-09 15:10 +0000 [r121231] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 121230 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500 - (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that - this is being merged to trunk/1.6.0 because it may affect - non-callback agents with ackcall set) ........ r121229 | - mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 - lines A unique situation of timeouts brought forth a failure - situation for autologoff in chan_agent. If using - AgentCallbackLogin-style agents, then if the timeout specified by - the Dial() to reach the agent's phone was shorter than the - timeout specified in queues.conf, then autologoff would only work - if the caller hung up while the agent's phone was ringing. This - patch allows autologoff to work in this situation when the call - in queue transfers to the next available agent (as it would have - if the timeout in queues.conf were less than the timeout in the - Dial()). (closes issue #12754) Reported by: Rodrigo Patches: - 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo - ........ ................ - -2008-06-08 01:43 +0000 [r121138-121164] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_console.c: Merged revisions 121163 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) - | 4 lines This was accidentally reverted. Fixes a bug where if a - stream monitor thread was not created (caused from failure of - opening or starting the stream) pthread_cancel was called with an - invalid thread ID. ........ - - * apps/app_parkandannounce.c, /: Merged revisions 121131 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 - Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce. - Also, loop made more efficient as announce template only needs to - be checked until the number of colon separated arguments run out, - not the entire pointer storage array. Was done in a similiar - fashion in 1.4, but here we're using less variables. ........ - -2008-06-07 14:19 +0000 [r121080] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions - 121079 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008) - | 15 lines Merged revisions 121078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) - | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes - issue #12807) Reported by: ys Patches: chan_agent_local.diff - uploaded by ys (license 281) ........ ................ - -2008-06-06 20:25 +0000 [r121011-121047] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 121010 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 | - tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines - Make extension match characters case-insensitive. (closes issue - #12777) Reported by: jsmith Patches: - lower_case_patterns-trunk-v1.patch uploaded by jsmith (license - 15) ........ - -2008-06-06 18:31 +0000 [r120907-120961] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008) - | 9 lines Merged revisions 120959 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) - | 1 line add another LOW_MEMORY define I forgot ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008) - | 9 lines Merged revisions 120908 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) - | 1 line only define thread storage variable if necessary for - LOW_MEMORY ........ ................ - - * channels/chan_sip.c: Merged revisions 120906 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) - | 16 lines Merged revisions 120863,120885 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) - | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two - allocations of the ast_rtp struct that were previously allocated - on the stack have been modified to use thread local storage - instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 - (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make - sure proper destructor function is called as well define two - thread storage local variables. ........ ................ - -2008-06-06 17:35 +0000 [r120864-120905] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_exec.c: Merged revisions 120904 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 | - tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines - For the purpose of making the changed syntax to ExecIf easier to - transition, allow the deprecated syntax (fixed for jmls on -dev). - ........ - -2008-06-05 21:39 +0000 [r120829] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 120828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 | - murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a - small fix for a crash that occurs when compiling AEL with global - vars ........ - -2008-06-05 17:17 +0000 [r120677] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 120676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008) - | 10 lines Merged revisions 120675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) - | 2 lines Ignore appended resource when comparing JIDs. ........ - ................ - -2008-06-05 16:42 +0000 [r120643-120674] Brett Bryant <bbryant@digium.com> - -2008-06-05 16:01 +0000 [r120566-120603] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged - revisions 120602 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 | - tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines - Conditionally load the AGI command gosub, depending on whether or - not res_agi has been loaded, fix a return value in the loader, - and ensure that the help workhorse header does not print on load. - ........ - - * /, UPGRADE.txt: Merged revisions 120567 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 | - tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines - Add info on the [compat] section of asterisk.conf. ........ - - * apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793) - -2008-06-04 22:08 +0000 [r120515] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 120514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun - 2008) | 14 lines Merged revisions 120513 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun - 2008) | 6 lines Make sure that the string we set will survive the - unref of the queue member. Thanks to Russell, who pointed this - out. ........ ................ - -2008-06-04 20:35 +0000 [r120478] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 120477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 | - tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines - MSet doesn't necessarily need chan to be set ........ - -2008-06-04 15:38 +0000 [r120338] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 | - file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We - like tabs. ........ - -2008-06-04 14:13 +0000 [r120287] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 120286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun - 2008) | 15 lines Merged revisions 120285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun - 2008) | 7 lines Tab completion when removing a member should give - the member's interface, not the name, since the interface is what - is expected for the command. (closes issue #12783) Reported by: - davevg ........ ................ - -2008-06-04 13:34 +0000 [r120284] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) | - 14 lines Merged revisions 120282 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 - lines Fix a log message and add a message for when the dialplan - is done reloading. (closes issue #12716) Reported by: chappell - Patches: dialplan_reload_2.diff uploaded by chappell (license 8) - ........ ................ - -2008-06-03 23:18 +0000 [r120228-120234] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) - | 16 lines Merged revisions 120226 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) - | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD() - macro the loopback switch cannot perform any translation on the - extension number before searching for it in the target context. - (closes issue #12473) Reported by: chappell Patches: - pbx_loopback.c.diff uploaded by chappell (license 8) ........ - ................ - -2008-06-03 22:18 +0000 [r120178] Jeff Peeler <jpeeler@digium.com> - - * main/config.c: Merged revisions 120174 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) - | 14 lines Merged revisions 120173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) - | 6 lines (closes issue #11594) Reported by: yem Tested by: yem - This change decreases the buffer size allocated on the stack - substantially in config_text_file_load when LOW_MEMORY is turned - on. This change combined with the fix from revision 117462 - (making mkintf not copy the zt_chan_conf structure) was enough to - prevent the crash. ........ ................ - -2008-06-03 22:08 +0000 [r120172] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/options.h, main/asterisk.c, Makefile, - main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c, - configs/pbx_realtime.conf (removed): Merged revisions 120171 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 - Jun 2008) | 5 lines Move compatibility options into - asterisk.conf, default them to on for upgrades, and off for new - installations. This includes the translation from pipes to commas - for pbx_realtime and the EXEC command for AGI, as well as the - change to the Set application not to support multiple variables - at once. ........ - -2008-06-03 21:35 +0000 [r120170] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r120169 | russell | 2008-06-03 16:35:11 -0500 - (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) - | 4 lines Fix another place where peer->callno could change at a - very bad time, and also fix a place where a peer was used after - the reference was released. (inspired by rev 120001) ........ - ................ - -2008-06-03 16:24 +0000 [r120034] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500 - (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) - | 9 lines Save the callno when we're poking, because our peer - structure could change during destruction (and thus we unlock the - wrong callno, causing a cascade failure). (closes issue #12717) - Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded - by Corydon76 (license 14) Tested by: gewfie ........ - ................ - -2008-06-03 15:57 +0000 [r119931-120000] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-vtest21, - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ref.ael-vtest17, - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, - pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, - 03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 - lines Updated the regressions on AEL. Hadn't updated this for the - changes I made to preserve ${EXTEN} in switches, which affected - several tests because it adds extra priorities, and at least one - needed to be updated because of the removal of the empty - extension warning message. ........ ................ - - * res/ael/pval.c, /: Merged revisions 119930 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | - 24 lines Merged revisions 119929 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | - 16 lines as per - http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, - which is a message from Philipp Kempgen, requesting that the - WARNING that an extension is empty be reduced to a NOTICE or - less, as empty extensions are syntactically possible, and no big - deal. With which I agree, and have removed that WARNING message - entirely. I think it is not necessary to see this message. It - didn't state that a NoOp() was inserted automatically on your - behalf, and really, as users, who cares? Why freak out dialplan - writers with unnecessary warnings? The details of the - machinations a compiler goes thru to produce working assembly - code is of little interest to most programmers-- we will follow - the unix principal of doing our work silently. ........ - ................ - -2008-06-03 14:48 +0000 [r119928] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) | - 10 lines Merged revisions 119926 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 - lines Treat ECONNREFUSED as an error that will stop further - retransmissions. (issue #AST-58, patch from Switchvox) ........ - ................ - -2008-06-03 13:30 +0000 [r119745-119893] Russell Bryant <russell@digium.com> - - * /, main/logger.c: Merged revisions 119892 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 | - russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines - Do a deep copy of file and function strings to avoid a potential - crash when modules are unloaded. (closes issue #12780) Reported - by: ys Patches: logger.diff uploaded by ys (license 281) -- - modified by me for coding guidelines ........ - - * /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119839 | russell | 2008-06-02 15:08:24 -0500 - (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) - | 7 lines Revert a change made for issue #12479. This change - caused a regression such that a dial string such as (IAX2/foo) - did not automatically fall back to dialing the 's' extension - anymore. (closes issue #12770) Reported by: dagmoller ........ - ................ - - * /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) - | 4 lines Add app_fax from asterisk-addons, with some additional - changes to resolve compiler warnings, as well as update to the - APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under - the LGPL, so we can move this module into the main tree. ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 119799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 | - russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines - After determining that the version of spandsp installed is an - acceptable version, do a build and link test to ensure that the - library is usable, and that libtiff is also available ........ - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: - Merged revisions 119795 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 | - russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines - Add a configure script check for spandsp ........ - - * main/manager.c, /: Merged revisions 119744 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) - | 13 lines Merged revisions 119742 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) - | 5 lines Improve CLI command blacklist checking for the command - manager action. Previously, it did not handle case or whitespace - properly. This made it possible for blacklisted commands to get - executed anyway. (closes issue #12765) ........ ................ - -2008-06-02 14:40 +0000 [r119743] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c, - res/res_jabber.c: Merged revisions 119741 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 | - phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 - lines Do not link the guest account with any configured XMPP - client (in jabber.conf). The actual connection is made when a - call comes in Asterisk. Apply this fix to Jingle too. Fix the - ast_aji_get_client function that was not able to retrieve an XMPP - client from its JID. (closes issue #12085) Reported by: junky - Tested by: phsultan ........ - -2008-06-02 12:32 +0000 [r119532-119690] Russell Bryant <russell@digium.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged - revisions 119586,119637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) - | 9 lines Merged revisions 119585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) - | 1 line Added counter for unhandled_bmsg Print, this prevents - the logs to be flooded to fast and save CPU in this error - scenario. Added 'last_used' element to bc structure, when a - bchannel changes from used to free this exact time will be marked - in last_used. When a new channel is requested the find_free_chan - function will check if the new empty channel was used within the - last second, if yes it will search for the next channel, if no it - will return this channel. This simple mechanism has prooven to - prevent race conditions where the NT and TE tried to allocate the - exact same channel at the same time (RELEASE cause: 44). ........ - ................ r119637 | crichter | 2008-06-02 04:35:04 -0500 - (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 - Jun 2008) | 1 line fixed compile issue when dev-mode is enabled - ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119688 | russell | 2008-06-02 07:30:42 -0500 - (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) - | 3 lines Even of the first PING or LAGRQ doesn't get sent - because it comes up too soon, make sure to reschedule so it gets - sent later. ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119534 | russell | 2008-06-01 20:08:16 -0500 - (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) - | 2 lines Change a debug message to an actual debug message - ........ ................ - - * apps/app_dial.c, /: Merged revisions 119531 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) - | 10 lines Merged revisions 119530 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) - | 2 lines Fix another typo in documentation ........ - ................ - -2008-06-01 21:59 +0000 [r119529] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_dial.c, /: Merged revisions 119479 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) - | 10 lines Merged revisions 119478 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) - | 2 lines small typo fix 'retires' => 'retries' ........ - ................ - -2008-05-30 21:24 +0000 [r119420] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 119419 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008) - | 14 lines Merged revisions 119404 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) - | 6 lines When joinempty=strict, it only failed on join if there - were busy members. If all members were logged out OR paused, then - it (incorrectly) let callers join the queue. (closes issue - #12451) Reported by: davidw ........ ................ - -2008-05-30 19:48 +0000 [r119356] Joshua Colp <jcolp@digium.com> - - * main/autoservice.c, /: Merged revisions 119355 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) | - 10 lines Merged revisions 119354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 - lines Fix a bug I found while testing for another issue. ........ - ................ - -2008-05-30 17:13 +0000 [r119304] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Oops, broke 1.6 (thanks MattF) - -2008-05-30 16:57 +0000 [r119303] Michiel van Baak <michiel@vanbaak.info> - - * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, - contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.mandrake.asterisk, /, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.gentoo.asterisk, - contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200 - (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) - | 14 lines dont use a bashism way to check the $VERSION variable. - The rc/init.d scripts, and safe_asterisk work on normal sh now - again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy - (me and loloski) FC9 (loloski) (closes issue #12687) Reported by: - loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt - uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak - ........ ................ - -2008-05-30 16:40 +0000 [r119297-119300] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c, /: Merged revisions 119299 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 | - tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines - Suppress warning about pbx structure already existing ........ - - * apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /, - CHANGES: Merged revisions 119296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | - tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines - Add native AGI command GOSUB, as invoking Gosub with EXEC does - not work properly. (closes issue #12760) Reported by: Corydon76 - Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 - (license 14) Tested by: tim_ringenbach, Corydon76 ........ - -2008-05-30 13:01 +0000 [r119158-119240] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119239 | russell | 2008-05-30 07:59:11 -0500 - (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r119238 | russell | 2008-05-30 07:55:36 -0500 - (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) - | 7 lines - Instead of only enforcing destination call number - checking on an ACK, check all full frames except for PING and - LAGRQ, which may be sent by older versions too quickly to contain - the destination call number. (As suggested by Tim Panton on the - asterisk-dev list) - Merge changes from - team/russell/iax2-frame-race, which prevents PING and LAGRQ from - being sent before the destination call number is known. ........ - ................ ................ - - * main/autoservice.c, /: Merged revisions 119157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008) - | 18 lines Merged revisions 119156 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) - | 10 lines Fix a race condition in channel autoservice. There was - still a small window of opportunity for a DTMF frame, or some - other deferred frame type, to come in and get dropped. (closes - issue #12656) (closes issue #12656) Reported by: dimas Patches: - v3-12656.patch uploaded by dimas (license 88) -- with some - modifications by me ........ ................ - -2008-05-29 20:26 +0000 [r119073] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008) - | 15 lines Merged revisions 119071 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) - | 7 lines Call waiting tone occurs too often, because it's - getting serviced by both subchannels. (closes issue #11354) - Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded - by Corydon76 (license 14) ........ ................ - -2008-05-29 19:06 +0000 [r118960-119014] Russell Bryant <russell@digium.com> - - * apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119013 | russell | 2008-05-29 14:05:33 -0500 - (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) - | 4 lines - Fix a typo in the argument to Playtones - use - ast_safe_sleep() instead of calling the wait application (thanks - to tilghman for pointing these out!) ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r119010 | russell | 2008-05-29 13:54:11 -0500 - (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r119009 | russell | 2008-05-29 13:49:12 -0500 - (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) - | 7 lines Merge changes from - team/russell/iax2-another-fix-to-the-fix As described in the - following post to the asterisk-dev mailing list, only enforce - destination call numbers when processing an ACK. - http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html - (closes issue #12631) ........ ................ ................ - - * apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r118962 | russell | 2008-05-29 12:52:00 -0500 - (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) - | 3 lines - Mark app_milliwatt dependent on res_indications - (thanks to jsmith) - fix a typo in a log message (thanks to - qwell) ........ ................ - - * apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r118959 | russell | 2008-05-29 12:46:04 -0500 - (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008) - | 3 lines Change milliwatt to use the proper tone by default - (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz - for anyone that might want it. ........ ................ - -2008-05-29 17:42 +0000 [r118958] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_mgcp.c, channels/chan_zap.c, /, - channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, - include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions - 118955,118957 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) - | 11 lines Merged revisions 118953 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) - | 3 lines Add some debugging code that ensures that when we do - deadlock avoidance, we don't lose the information about how a - lock was originally acquired. ........ ................ r118957 | - tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 - lines Merged revisions 118954 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) - | 2 lines Define also when not DEBUG_THREADS ........ - ................ - -2008-05-29 04:11 +0000 [r118909] Steve Murphy <murf@digium.com> - - * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed, - 28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | - 46 lines (closes issue #10668) (closes issue #11721) (closes - issue #12726) Reported by: arkadia Tested by: murf These changes: - 1. revert the changes made via bug 10668; I should have known - that such changes, even tho they made sense at the time, seemed - like an omission, etc, were actually integral to the CDR system - via forkCDR. It makes sense to me now that forkCDR didn't - natively end any CDR's, but rather depended on natively closing - them all at hangup time via traversing and closing them all, - whether locked or not. I still don't completely understand the - benefits of setvar and answer operating on locked cdrs, but I've - seen enough to revert those changes also, and stop messing up - users who depended on that behavior. bug 12726 found reverting - the changes fixed his changes, and after a long review and - working on forkCDR, I can see why. 2. Apply the suggested - enhancements proposed in 10668, but in a completely compatible - way. ForkCDR will behave exactly as before, but now has new - options that will allow some actions to be taken that will - slightly modify the outcome and side-effects of forkCDR. Based on - conversations I've had with various people, these small tweaks - will allow some users to get the behavior they need. For - instance, users executing forkCDR in an AGI script will find the - answer time set, and DISPOSITION set, a situation not covered - when the routines were first written. 3. A small problem in the - cdr serializer would output answer and end times even when they - were not set. This is now fixed. ........ ................ - -2008-05-28 18:07 +0000 [r118781] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008) - | 2 lines remove unused astobj.h header file from chan_skinny.c - ........ - -2008-05-28 14:31 +0000 [r118648] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged - revisions 118647 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | - 12 lines Merged revisions 118646 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 - lines Add an option to use the source IP address of RTP as the - destination IP address of UDPTL when a specific option is - enabled. If the remote side is properly configured (ports - forwarded) then UDPTL will flow. (closes issue #10417) Reported - by: cstadlmann ........ ................ - -2008-05-28 14:13 +0000 [r118615-118645] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged - revisions 118644 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 | - phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 - lines Changed to temporary namespaces to match with latest XEPs. - As soon as Jingle is completely standardized, we can set those - namespaces to their final values. Added two attributes to the - jingle_pvt struct to store the content name attributes. Reported - by Robert McQueen on Telepathy's framework mailing list : - http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html - Keeping working on our Jingle stack! ........ - - * channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) - | 1 line Code simplification ........ - -2008-05-27 19:35 +0000 [r118561] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | - 12 lines Merged revisions 118558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 - lines Fix an issue where codec preferences were not set on - dialogs that were not authenticated via a user or peer and allow - framing to work without rtpmap in the SDP. (closes issue #12501) - Reported by: slimey ........ ................ - -2008-05-27 19:28 +0000 [r118557] Russell Bryant <russell@digium.com> - - * /, include/asterisk/compat.h: Merged revisions 118556 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27 - May 2008) | 6 lines Add printf format attribute for vasprintf(). - (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff - uploaded by snuffy (license 35) ........ - -2008-05-27 19:22 +0000 [r118555] Tilghman Lesher <tlesher@digium.com> - - * main/cli.c, /: Merged revisions 118554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008) - | 14 lines Merged revisions 118551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) - | 6 lines When showing an error message for a command, don't - shorten the command output, as it tends to confuse the user (it's - fine for suggesting other commands, however). Reported by: - seanbright (on #asterisk-dev) Fixed by: me ........ - ................ - -2008-05-27 19:09 +0000 [r118518] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May - 2008) | 19 lines Merged revisions 118509 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May - 2008) | 11 lines Russell noted to me that in the case that - separate threads use their own addressing system, the fix I made - for issue 12376 does not guarantee uniqueness to the datastores' - uids. Though I know of no system that works this way, I am going - to change this right now to prevent trying to track down some - future bug that may occur and cause untold hours of debugging - time to track down. The change involves using a global counter - which increases with each new chanspy_ds which is created. This - guarantees uniqueness. ........ ................ - -2008-05-27 18:59 +0000 [r118471] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 118466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008) - | 16 lines Merged revisions 118465 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) - | 8 lines NULL character should terminate only commands back to - the core, not log messages to the console. (closes issue #12731) - Reported by: seanbright Patches: 20080527__bug12731.diff.txt - uploaded by Corydon76 (license 14) Tested by: seanbright ........ - ................ - -2008-05-27 17:25 +0000 [r118418] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_voicemail.c: small update to the g() option of - app_voicemail to note that gain changes only work on zap channels - right now. issue #12578 shows it's not clear right now. - -2008-05-27 16:48 +0000 [r118378-118382] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May - 2008) | 22 lines Merged revisions 118365 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May - 2008) | 14 lines Add a unique id to the datastore allocated in - app_chanspy since it is possible that multiple spies may be - listening to the same channel. (closes issue #12376) Reported by: - DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut - (license 60) Tested by: destiny6628 (closes issue #12243) - Reported by: atis ........ ................ - - * /: Hmm, I apparently forgot to commit the block of revision - 118175. Now I'm doing it. - -2008-05-27 15:47 +0000 [r118360] Tilghman Lesher <tlesher@digium.com> - - * /, configs/queues.conf.sample: Merged revisions 118359 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500 - (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) - | 3 lines Add a note that pbx_config.so is needed for Local - channels. (Closes issue #12671) ........ ................ - -2008-05-27 14:51 +0000 [r118331] Russell Bryant <russell@digium.com> - - * /, include/asterisk/compat.h: Merged revisions 118328 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27 - May 2008) | 2 lines Add printf attribute to asprintf ........ - -2008-05-27 13:30 +0000 [r118301-118303] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008) - | 6 lines When binding anonymously, credentials are still needed. - (closes issue #12601) Reported by: suretec Patches: - res_config_ldap.c.patch uploaded by suretec (license 70) ........ - - * /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 | - tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines - In compat14 mode, don't translate pipes inside expressions, as - they aren't argument delimiters, but rather 'or' symbols. (Closes - issue #12723) ........ - -2008-05-25 16:20 +0000 [r118253] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) - | 20 lines Merged revisions 118251 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) - | 12 lines Realtime flag affects construction in multiple ways, - so consulting whether rtcachefriends was set was done too soon - (needed to be done inside build_peer, not just as a flag to - build_peer). Also, fullcontact needed to be reconstructed, - because realtime separates the embedded ';' into multiple fields. - (closes issue #12722) Reported by: barthpbx Patches: - 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) - Tested by: barthpbx (Much of the discussion happened on - #asterisk-dev for diagnosing this issue) ........ - ................ - -2008-05-24 01:15 +0000 [r118177-118179] Jeff Peeler <jpeeler@digium.com> - - * doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23 - May 2008) | 1 line add document describing API changes from 1.4.0 - to 1.6.0 ........ - -2008-05-23 21:37 +0000 [r118168] Brett Bryant <bbryant@digium.com> - - * main/manager.c, /, main/http.c, include/asterisk/manager.h: - Merged revisions 118161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 | - bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines - Add new functionality to http server that requires manager - authentication for any path that includes a directory named - 'private'. This patch also requires manager authentication for - any POST's being sent to the server as well to help secure - uploads. ........ - -2008-05-23 21:31 +0000 [r118165] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Merged revisions 118164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008) - | 9 lines Merged revisions 118163 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) - | 1 line Fix a few things I missed to ensure zt_chan_conf - structure is not modified in mkintf ........ ................ - -2008-05-23 18:15 +0000 [r118130] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, /: Merged revisions 118129 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 | - tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines - Protect the object from changing while the 'odbc show' CLI - command is running (Closes issue #12704) ........ - -2008-05-23 13:00 +0000 [r118054] Tilghman Lesher <tlesher@digium.com> - - * doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008) - | 11 lines Merged revisions 118052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) - | 3 lines Add information on using the Asterisk console, - including tab command line completion. (Closes issue #12681) - ........ ................ - -2008-05-23 12:37 +0000 [r118050] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h, /, main/utils.c: Merged revisions - 118049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008) - | 17 lines Merged revisions 118048 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) - | 9 lines Don't declare a function that takes variable arguments - as inline, because it's not valid, and on some compilers, will - emit a warning. - http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes - issue #12289) Reported by: francesco_r Patches by Tilghman, final - patch by me ........ ................ - -2008-05-23 11:02 +0000 [r118021] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions - 118020 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 | - phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 - lines - remove whitespaces between tags in received XML packets - before giving them to the parser ; - report Gtalk error messages - from a buddy to the console. This patch makes Asterisk "Google - Jingle" (chan_gtalk) implementation work with Empathy. Note that - this is only true for audio streams, not video. Thank you to PH - for his great help! (closes issue #12647) Reported by: PH - Patches: trunk-12647-1.diff uploaded by phsultan (license 73) - Tested by: phsultan, PH ........ - -2008-05-22 21:43 +0000 [r117984-117987] Tilghman Lesher <tlesher@digium.com> - - * /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged - revisions 117986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 | - tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines - Add a compatibility option for upgrading realtime extensions - ........ - -2008-05-22 18:55 +0000 [r117901] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 117900 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008) - | 10 lines Merged revisions 117899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) - | 2 lines Also remove preamble from asynchronous events (reported - by jsmith on #asterisk-dev) ........ ................ - -2008-05-22 15:51 +0000 [r117793] Sean Bright <sean.bright@gmail.com> - - * /, configs/jabber.conf.sample: Merged revisions 117792 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, - 22 May 2008) | 1 line Minor text fix. roster -> resource. - ........ - -2008-05-22 13:41 +0000 [r117757] Russell Bryant <russell@digium.com> - - * main/asterisk.c, /, build_tools/make_buildopts_h: Merged - revisions 117756 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 | - russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines - Store build-time options as a string in AST_BUILDOPTS in - buildopts.h. Also, display this information in the "core show - settings" CLI command. This is useful if you want to verify that - you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc. - ........ - -2008-05-21 22:01 +0000 [r117659-117660] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: Merged revisions 117658 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008) - | 10 lines Merged revisions 117582 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) - | 2 lines Ensure that passed in zt_chan_conf structure is not - modified in mkintf. ........ ................ - - * channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008) - | 12 lines Merged revisions 117462 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) - | 3 lines Pass a pointer for the conf parameter to the function - mkintf rather than the whole zt_chan_conf structure. Another - commit is following to make sure the zt_chan_conf structure is - not modified. ........ ................ - -2008-05-21 19:45 +0000 [r117576] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) | - 10 lines Merged revisions 117574 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 - lines Apply the autoframing setting to dialogs that do not get - matched against a user or peer. ........ ................ - -2008-05-21 18:44 +0000 [r117522] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 117520 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008) - | 11 lines Merged revisions 117519 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) - | 3 lines Strip the preamble from the output also when -rx is not - being used (Related to issue #12702) ........ ................ - -2008-05-21 18:29 +0000 [r117486-117516] Russell Bryant <russell@digium.com> - - * main/asterisk.c, /: Merged revisions 117515 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008) - | 12 lines Merged revisions 117514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) - | 4 lines Don't filter the magic character in the network - verboser. It gets filtered once it reaches the client. (related - to issue #12702, pointed out by tilghman) ........ - ................ - - * main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r117508 | russell | 2008-05-21 13:20:11 -0500 - (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) - | 7 lines 1) Don't print the verbose marker in front of every - message from ast_verbose() being sent to remote consoles. 2) Fix - pbx_gtkconsole to filter out the verbose marker. (related to - issue #12702) ........ ................ - - * main/asterisk.c, /: Merged revisions 117481 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008) - | 14 lines Merged revisions 117479 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) - | 6 lines Don't display the verbose marker for calls to - ast_verbose() that do not include a VERBOSE_PREFIX in front of - the message. (closes issue #12702) Reported by: johnlange Patched - by me ........ ................ - -2008-05-21 02:21 +0000 [r117368] Mark Michelson <mmichelson@digium.com> - - * main/config.c, /: Merged revisions 117367 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 | - mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19 - lines Be sure that we cache included files for each source file - which loads a configuration file. As it was, only the first did - so. This led to a problem if the included file was changed (but - not the configuration file which includes it) and the second - source file attempted to reload the configuration. It would not - see that the included file had changed. In this particular - example, res_phoneprov and chan_sip both loaded sip.conf, which - included a file call sip.peers.conf. Since res_phoneprov was the - first to load sip.conf, only it cached the fact that sip.conf - included sip.peers.conf. If sip.peers.conf were changed and - sip.conf were not and a sip reload were issued (meaning that - chan_sip attempts to reload sip.conf only if it and its included - files have changed) the changes made to sip.peers.conf would not - be seen and therefore no action would be taken. (closes issue - #12693) Reported by: marsosa ........ - -2008-05-21 01:20 +0000 [r117365] Steve Murphy <murf@digium.com> - - * /, utils/ael_main.c: Merged revisions 117335 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 | - murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines - These changes were made via the comments atis_work made at 4:30am - (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He - noted that a backslash was being inserted before commas in app - call arguments in the extensions.conf.aeldump file that you get - from aelparse with the -w arg. This was being generated from code - left over from 1.4, where commas were substituted with '|', and - any remaining commas needed to be escaped. Many thanks to atis - for his comment; please let us know if these changes break - anything! ........ - -2008-05-19 16:58 +0000 [r117134-117137] Joshua Colp <jcolp@digium.com> - - * res/res_smdi.c, /: Merged revisions 117136 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) | - 14 lines Merged revisions 117135 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 - lines Use the right pthread lock and condition when waiting. - (closes issue #12664) Reported by: tomo1657 Patches: - res_smdi.c.patch uploaded by tomo1657 (license 484) ........ - ................ - -2008-05-19 16:07 +0000 [r117089] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500 - (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) - | 2 lines The addition of usleep(2) within ast_assert requires - the inclusion of the unistd.h header ........ ................ - -2008-05-19 16:05 +0000 [r117083-117087] Joshua Colp <jcolp@digium.com> - - * /, main/logger.c: Merged revisions 117085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 | - file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The - logger closes the files it is logging to when reloading so we - have to read in the logger configuration even if it has not - changed so that the logs get opened again. (closes issue #12665) - Reported by: DennisD ........ - - * /, channels/h323/ast_h323.cxx: Merged revisions 117082 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon, - 19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 - lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) - Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby - (license 430) ........ ................ - -2008-05-19 03:44 +0000 [r116980] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r116979 | russell | 2008-05-18 22:44:28 -0500 - (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) - | 4 lines Avoid access of uninitialized memory. This caused a - bunch of crashes for me while doing load testing of development - branch where I'm working on some performance improvements. - ........ ................ - -2008-05-18 21:18 +0000 [r116949] Tilghman Lesher <tlesher@digium.com> - - * /, utils/astcanary.c: Merged revisions 116948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 | - tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines - Add a set of text to the file astcanary uses to communicate back - the main Asterisk process, which explains the purpose for the - file being there. This should assist people who find the file and - wonder why it exists. ........ - -2008-05-18 19:59 +0000 [r116922] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 | - russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines - Remove duplicate colon on Reason header (closes issue #12678) - ........ - -2008-05-17 19:40 +0000 [r116849-116885] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri, - 16 May 2008) | 12 lines Merged revisions 116799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 - lines Check to make sure an RTP structure exists before calling - ast_rtp_new_source on it. (closes issue #12669) Reported by: - sbisker ........ ................ - -2008-05-16 20:03 +0000 [r116798] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 | - mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try - to see if we can make our ringback situation a little better - ........ - -2008-05-15 22:07 +0000 [r116636-116695] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/utils.h, /, include/asterisk/strings.h: Merged - revisions 116694 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 | - tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines - Add an extra check in ast_strlen_zero, and make ast_assert() not - print the file, line, and function name twice. (Closes issue - #12650) ........ - - * cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 | - tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines - Don't unload config on reload, when config has not changed. - (Closes issue #12652) ........ - -2008-05-14 21:41 +0000 [r116470] Russell Bryant <russell@digium.com> - - * main/rtp.c, main/sched.c, main/channel.c, main/udptl.c, - include/asterisk/utils.h, /, channels/chan_agent.c, - main/abstract_jb.c, include/asterisk/channel.h: Merged revisions - 116469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) - | 12 lines Merged revisions 116463 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) - | 4 lines Add ast_assert(), which can be used to handle fatal - errors. It is only compiled in if dev-mode is enabled, and only - aborts if DO_CRASH is defined. (inspired by issue #12650) - ........ ................ - -2008-05-14 21:39 +0000 [r116468] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 116467 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) - | 15 lines Merged revisions 116466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) - | 7 lines Avoid zombies when the channel exits before the AGI. - (closes issue #12648) Reported by: gkloepfer Patches: - 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14) - Tested by: gkloepfer ........ ................ - -2008-05-14 20:43 +0000 [r116408-116411] Jason Parker <jparker@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 116410 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r116410 | qwell | 2008-05-14 15:43:26 -0500 - (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May - 2008) | 1 line Document exitcontext in app_voicemail sample - config ........ ................ - - * apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | - 9 lines Voicemail "* exit" should not require an exitcontext to - be specified. The behavior in 1.4 was that it would use the - current context if an exitcontext existed. (closes issue #12605) - Reported by: kenjreno Patches: 12605-starexit.diff uploaded by - qwell (license 4) Tested by: file ........ - -2008-05-14 18:54 +0000 [r116351-116354] Joshua Colp <jcolp@digium.com> - - * /, main/Makefile: Merged revisions 116353 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) | - 12 lines Merged revisions 116352 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 - lines Add linux-gnueabi in. (closes issue #12529) Reported by: - tzafrir ........ ................ - - * /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4 - lines Make the ldap version setting work without having both - version and protocol set. (closes issue #12613) Reported by: - suretec ........ - -2008-05-14 17:01 +0000 [r116319] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500 - (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) - | 2 lines Detect another way for a connection to have gone away. - (closes issue #12618) Reported by: ctooley Patches: - 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136) - trunk-externalivr-test_fd.diff uploaded by ctooley (license 136) - ........ ................ - -2008-05-14 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta9 released. - -2008-05-14 13:13 +0000 [r116236] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11 - lines Merged revisions 116230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 - lines Accept text messages even with Content-Type: - text/plain;charset=Södermanländska ........ ................ - -2008-05-14 00:20 +0000 [r116096-116139] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /, include/asterisk/lock.h: Merged revisions - 116089 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May - 2008) | 20 lines Merged revisions 116088 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May - 2008) | 12 lines A change to the way channel locks are handled - when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, - it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in - menuselect, the actual origin of channel locks is obscured by the - fact that all channel locks appear to happen in the function - ast_channel_lock(). This code change redefines ast_channel_lock - to be a macro which maps to __ast_channel_lock(), which then - relays the proper file name, line number, and function name - information to the core lock functions so that this information - will be displayed in the case that there is some sort of locking - error or core show locks is issued. ........ ................ - -2008-05-13 21:19 +0000 [r116020-116040] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, /: Merged revisions 116039 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r116039 | russell | 2008-05-13 16:18:55 -0500 - (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) - | 24 lines Fix a deadlock involving channel autoservice and - chan_local that was debugged and fixed by mmichelson and me. We - observed a system that had a bunch of threads stuck in - ast_autoservice_stop(). The reason these threads were waiting - around is because this function waits to ensure that the channel - list in the autoservice thread gets rebuilt before the stop() - function returns. However, the autoservice thread was also - locked, so the autoservice channel list was never getting - rebuilt. The autoservice thread was stuck waiting for the channel - lock on a local channel. However, the local channel was locked by - a thread that was stuck in the autoservice stop function. It - turned out that the issue came down to the local_queue_frame() - function in chan_local. This function assumed that one of the - channels passed in as an argument was locked when called. - However, that was not always the case. There were multiple cases - in which this channel was not locked when the function was - called. We fixed up chan_local to indicate to this function - whether this channel was locked or not. The previous assumption - had caused local_queue_frame() to improperly return with the - channel locked, where it would then never get unlocked. (closes - issue #12584) (related to issue #12603) ........ ................ - - * main/autoservice.c, /: Merged revisions 116001 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) - | 13 lines Merged revisions 115990 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) - | 5 lines Fix an issue that I noticed in autoservice while - mmichelson and I were debugging a different problem. I noticed - that it was theoretically possible for two threads to attempt to - start the autoservice thread at the same time. This change makes - the process of starting the autoservice thread, thread-safe. - ........ ................ - -2008-05-13 20:30 +0000 [r115946] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, - 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 - lines Use the right flag to open the audio in non-blocking. - (closes issue #12616) Reported by: nicklewisdigiumuser ........ - ................ - -2008-05-13 20:19 +0000 [r115940-115942] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 | - mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line - Need to clear calling_party_cat variable after we retrieve it - ........ - - * channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 | - mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add - support for receiving calling party category ........ - -2008-05-13 18:38 +0000 [r115887] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 115886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) - | 11 lines Merged revisions 115884 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) - | 3 lines If the socket dies (read returns 0=EOF), return - immediately. (Closes issue #12637) ........ ................ - -2008-05-13 17:48 +0000 [r115848-115851] Russell Bryant <russell@digium.com> - - * res/res_smdi.c, /: Merged revisions 115847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 | - russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines - Initialize the start time in smdi_msg_wait. Somehow this code got - lost in trunk. ........ - -2008-05-12 17:57 +0000 [r115738] Mark Michelson <mmichelson@digium.com> - - * main/utils.c: Merged revisions 115737 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May - 2008) | 15 lines Merged revisions 115735 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May - 2008) | 7 lines If a thread holds no locks, do not print any - information on the thread when issuing a core show locks command. - This will help to de-clutter output somewhat. Russell said it - would be fine to place this improvement in the 1.4 branch, so - that's why it's going here too. ........ ................ - -2008-05-12 16:36 +0000 [r115706] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 115705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 | - qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line - Correctly document state interface for AddQueueMember. Discovered - while looking at issue #12626. ........ - -2008-05-12 15:18 +0000 [r115672] Brett Bryant <bbryant@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) - | 3 lines A small change to fix iax2 native bridging. ........ - -2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 115600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 | - mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add - Zap MTP2 support to chan_zap ........ - - * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 | - mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line - Open up audio channel when we get ACM on SS7 event ........ - -2008-05-10 14:22 +0000 [r115597] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 | - tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines - Ensure that "calldate" is acceptable for a column name. ........ - -2008-05-09 16:38 +0000 [r115581] Joshua Colp <jcolp@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 115580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) | - 10 lines Merged revisions 115579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 - lines Improve res_ninit and res_ndestroy autoconf logic on the - Darwin platform. ........ ................ - -2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115569 | russell | 2008-05-08 14:20:35 -0500 - (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) - | 2 lines Remove debug output. ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115566 | russell | 2008-05-08 14:17:04 -0500 - (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r115565 | russell | 2008-05-08 14:15:25 -0500 - (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) - | 25 lines Fix a race condition that bbryant just found while - doing some IAX2 testing. He was running Asterisk trunk running - IAX2 calls through a few Asterisk boxes, however, the audio was - extremely choppy. We looked at a packet trace and saw a storm of - INVAL and VNAK frames being sent from one box to another. It - turned out that what had happened was that one box tried to send - a CONTROL frame before the 3 way handshake had completed. So, - that frame did not include the destination call number, because - it didn't have it yet. Part of our recent work for security - issues included an additional check to ensure that frames that - are supposed to include the destination call number have the - correct one. This caused the frame to be rejected with an INVAL. - The frame would get retransmitted for forever, rejected every - time ... This race condition exists in all versions that got the - security changes, in theory. However, it is really only likely - that this would cause a problem in Asterisk trunk. There was a - control frame being sent (SRCUPDATE) at the _very_ beginning of - the call, which does not exist in 1.2 or 1.4. However, I am - fixing all versions that could potentially be affected by the - introduced race condition. These changes are what bbryant and I - came up with to fix the issue. Instead of simply dropping control - frames that get sent before the handshake is complete, the code - attempts to wait a little while, since in most cases, the - handshake will complete very quickly. If it doesn't complete - after yielding for a little while, then the frame gets dropped. - ........ ................ ................ - - * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) - | 11 lines Merged revisions 115561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) - | 3 lines Don't give up on attempting an outbound registration if - we receive a 408 Timeout. (closes issue #12323) ........ - ................ - - * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions - 115558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) - | 11 lines Merged revisions 115557 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) - | 3 lines remove postgres_cdr.sql, as the CDR schema is in - realtime_pgsql.sql, as well (closes issue #9676) ........ - ................ - - * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115555 | russell | 2008-05-08 10:32:48 -0500 - (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) - | 3 lines Don't exit the script if Asterisk is not running. - (closes issue #12611) ........ ................ - - * main/pbx.c, /: Merged revisions 115552 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) - | 12 lines Merged revisions 115551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) - | 4 lines Don't use a channel before checking for channel - allocation failure. (closes issue #12609) Reported by: edantie - ........ ................ - -2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 | - mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line - Remove unused code as well as demote an error message to a debug - message ........ - -2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant <russell@digium.com> - - * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115546 | russell | 2008-05-08 09:41:12 -0500 - (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) - | 4 lines Use the same method for executing Asterisk as the rest - of the script. (closes issue #12611) Reported by: b_plessis - ........ ................ - -2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant <russell@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) - | 6 lines Only save a password if a username exists. (closes - issue #12600) Reported By: suretec Patch by me ........ - - * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) - | 7 lines Use the default that the log output claims will be used - for the basedn (closes issue #12599) Reported by: suretec - Patches: 12599.patch uploaded by juggie (license 24) ........ - - * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) - | 2 lines Let chan_h323 build in dev mode ........ - - * /, include/asterisk/dlinkedlists.h (removed), - channels/chan_iax2.c: Merged revisions 115513 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) - | 19 lines Merged revisions 115512 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r115512 | russell | 2008-05-07 11:24:09 -0500 - (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) - | 3 lines Remove remnants of dlinkedlists. I didn't actually use - them in the final version of my IAX2 improvements. ........ - ................ ................ - -2008-05-07 13:49 +0000 [r115510] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/asterisk.ldap-schema, - contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 - May 2008) | 2 lines Update typos in description fields (closes - issue #12598) Reported by: suretec Patches: - asterisk_schema_changes.patch uploaded by suretec (license 70) - ........ - -2008-05-06 19:56 +0000 [r115420-115424] Jason Parker <jparker@digium.com> - - * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115423 | qwell | 2008-05-06 14:55:45 -0500 - (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r115422 | qwell | 2008-05-06 14:55:29 -0500 - (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | - 7 lines read requires an argument on some non-bash shells (closes - issue #12593) Reported by: bkruse Patches: - getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) - ........ ................ ................ - - * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115419 | qwell | 2008-05-06 14:38:44 -0500 - (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | - 7 lines Switch to using ast_random() rather than just rand(). - This does not fix the bug reported, but I believe it is correct. - (from issue #12446) Patches: bug_12446.diff uploaded by snuffy - (license 35) ........ ................ - -2008-05-06 19:33 +0000 [r115417] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 115416 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008) - | 10 lines Merged revisions 115415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) - | 2 lines Don't print the terminating NUL. (Closes issue #12589) - ........ ................ - -2008-05-06 13:57 +0000 [r115343] Joshua Colp <jcolp@digium.com> - - * /, configure, configure.ac: Merged revisions 115342 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue, - 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2 - lines Add in missing argument. ........ ................ - -2008-05-05 23:01 +0000 [r115335] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500 - (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) - | 7 lines Separate verbose output from CLI output, by using a - preamble. (closes issue #12402) Reported by: Corydon76 Patches: - 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 - (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt - uploaded by Corydon76 (license 14) ........ ................ - -2008-05-05 22:17 +0000 [r115331] Joshua Colp <jcolp@digium.com> - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, codecs/codec_speex.c, - configure.ac: Merged revisions 115328 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) | - 10 lines Merged revisions 115327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 - lines Make sure that either the main speex library contains - preprocess functions or that speexdsp does. If both fail then - speex stuff can not be built. ........ ................ - -2008-05-05 22:14 +0000 [r115330] Mark Michelson <mmichelson@digium.com> - - * main/config.c, /: Merged revisions 115329 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 | - mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15 - lines #execing the same file multiple times led to warning - messages saying that the same file was being #included twice. - This was due to the fact that #exec created a temporary file - which was then #included. The name of the temporary file was the - name of the #exec'd file, with the Unix timestamp and thread ID - concatenated. The issue was that if multiple #exec statements of - the same file were reached in the same second, then the result - was that the temporary files would have duplicate names. To - resolve this, the temporary file now has microsecond resolution - for the timestamp portion. (closes issue #12574) Reported by: - jmls Patches: 12574.patch uploaded by putnopvut (license 60) - Tested by: jmls, putnopvut ........ - -2008-05-05 21:44 +0000 [r115322] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May - 2008) | 21 lines Merged revisions 115320 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May - 2008) | 13 lines Don't consider a caller "handled" until the - caller is bridged with a queue member. There was too much of an - opportunity for the member to hang up (either during a delay, - announcement, or overly long agi) between the time that he - answered the phone and the time when he actually was bridged with - the caller. The consequence of this was that if the member hung - up in that interval, then proper abandonment details would not be - noted in the queue log if the caller were to hang up at any point - after the member hangup. (closes issue #12561) Reported by: - ablackthorn ........ ................ - -2008-05-05 20:28 +0000 [r115316] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) - | 2 lines Remove my rant, since I have now replaced the rant with - code. ........ - -2008-05-05 19:58 +0000 [r115310] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, /: Merged revisions 115309 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500 - (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) - | 2 lines Err, the documentation on the return value of - ast_odbc_backslash_is_escape is exactly backwards. ........ - ................ - -2008-05-05 19:50 +0000 [r115306] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) - | 13 lines Merged revisions 115304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) - | 5 lines Avoid putting opaque="" in Digest authentication. This - patch came from switchvox. It fixes authentication with Primus in - Canada, and has been in use for a very long time without causing - problems with any other providers. (closes issue AST-36) ........ - ................ - -2008-05-05 19:43 +0000 [r115303] Tilghman Lesher <tlesher@digium.com> - - * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 | - tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines - Note change for ExecIf syntax (caught by jmls on IRC) ........ - -2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming <kpfleming@digium.com> - - * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 | - kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2 - lines clarify wording ........ - -2008-05-05 03:26 +0000 [r115287] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, - contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.mandrake.asterisk, /, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.gentoo.asterisk, - contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500 - (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) - | 7 lines When starting Asterisk, bug out if Asterisk is already - running. (closes issue #12525) Reported by: explidous Patches: - 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14) - Tested by: mvanbaak ........ ................ - -2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp <jcolp@digium.com> - - * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat, - 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2 - lines Expand the test function for GCC attributes so that more - complex attributes are properly recognized. ........ - ................ - - * /, include/asterisk/compiler.h: Merged revisions 115280 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat, - 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 - lines For my next trick I will make these work with what our - autoconf header file gives us. ........ ................ - - * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat, - 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2 - lines Treat warnings as errors when checking if a GCC attribute - exists. We have to do this as GCC will just ignore the attribute - and pop up a warning, it won't actually fail to compile. ........ - ................ - -2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard <dhubbard@digium.com> - - * /: block voicemail mwi notification subscriptions taskprocessor - - * /: block pbx taskprocessor - - * /: block app_queue taskprocessor - - * /: blocked taskprocessors - -2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson <mmichelson@digium.com> - - * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500 - (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May - 2008) | 6 lines Clarify a comment that was, well, just wrong. It - turns out that ignoring the way that macros expand. Instead, I - have clarified in the comment why the macro will work even if the - scheduler id for the task to be deleted changes during the - execution of the macro. ........ ................ - -2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, /: Merged revisions 115104 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500 - (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) - | 2 lines Change the comment of deprecated to an actual compiler - deprecation ........ ................ - -2008-05-01 19:01 +0000 [r115020] Tilghman Lesher <tlesher@digium.com> - - * /, main/utils.c: Merged revisions 115018 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008) - | 14 lines Merged revisions 115017 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) - | 6 lines '#' is another reserved character for URIs that also - needs to be escaped. (closes issue #10543) Reported by: blitzrage - Patches: 20080418__bug10543.diff.txt uploaded by Corydon76 - (license 14) ........ ................ - -2008-05-01 17:28 +0000 [r114932] Russell Bryant <russell@digium.com> - - * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 | - russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines - Clarify the deprecation notice about Macro() to note that it will - not be removed for the sake of backwards compatibility, since it - is a non-trivial task to convert existing large dialplans that - depend on Macro() to use GoSub(), instead. ........ - -2008-05-01 16:52 +0000 [r114923] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 | - qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines - Allow dringXrange to properly default to 10, as was done in 1.4. - dringXrange is a new feature that was added, and it attempted to - default, but only when the option was specified. (closes issue - #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded - by qwell (license 4) Tested by: bjm ........ - - - -2008-04-30 20:20 +0000 [r114909] Russell Bryant <russell@digium.com> - - * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists - implementation from trunk - -2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Make 1.6.0 compile - -2008-04-30 17:06 +0000 [r114900] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 - lines Merged revisions 114890 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 - lines Don't crash on bad SIP replys. Fix created in Huntsville - together with Mark M (putnopvut) (closes issue #12363) Reported - by: jvandal Tested by: putnopvut, oej ........ ................ - -2008-04-30 16:41 +0000 [r114893] Russell Bryant <russell@digium.com> - - * /, channels/chan_console.c, channels/chan_iax2.c: Merged - revisions 114892 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) - | 36 lines Merged revisions 114891 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) - | 28 lines Merge changes from team/russell/iax2_find_callno and - iax2_find_callno_1.4 These changes address a critical performance - issue introduced in the latest release. The fix for the latest - security issue included a change that made Asterisk randomly - choose call numbers to make them more difficult to guess by - attackers. However, due to some inefficient (this is by far, an - understatement) code, when Asterisk chose high call numbers, - chan_iax2 became unusable after just a small number of calls. On - a small embedded platform, it would not be able to handle a - single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run - more than about 16 IAX2 channels. Ouch. These changes address - some performance issues of the find_callno() function that have - bothered me for a very long time. On every incoming media frame, - it iterated through every possible call number trying to find a - matching active call. This involved a mutex lock and unlock for - each call number checked. So, if the random call number chosen - was 20000, then every media frame would cause 20000 locks and - unlocks. Previously, this problem was not as obvious since - Asterisk always chose the lowest call number it could. A second - container for IAX2 pvt structs has been added. It is an astobj2 - hash table. When we know the remote side's call number, the pvt - goes into the hash table with a hash value of the remote side's - call number. Then, lookups for incoming media frames are a very - fast hash lookup instead of an absolutely insane array traversal. - In a quick test, I was able to get more than 3600% more IAX2 - channels on my machine with these changes. ........ - ................ - -2008-04-30 16:15 +0000 [r114889] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_console.c: Merged revisions 114888 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) - | 3 lines Fixes a bug where if a stream monitor thread was not - created (caused from failure of opening or starting the stream) - pthread_cancel was called with an invalid thread ID. ........ - -2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500 - (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr - 2008) | 2 lines use the ARRAY_LEN macro for indexing through the - iaxs/iaxsl arrays so that the size of the arrays can be adjusted - in one place, and change the size of the arrays from 32768 calls - to 2048 calls when LOW_MEMORY is defined ........ - ................ - - * /, Makefile.rules: Merged revisions 114876 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr - 2008) | 10 lines Merged revisions 114875 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr - 2008) | 2 lines pay attention to *all* header files for - dependency tracking, not just the local ones (inspired by r578 of - asterisk-addons by tilghman) ........ ................ - -2008-04-29 22:55 +0000 [r114867] Jeff Peeler <jpeeler@digium.com> - - * /, channels/iax2-provision.c: Merged revisions 114866 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 - Apr 2008) | 2 lines Fixes a problem where all the templates were - marked as dead no matter what. The templates should only be - marked as dead if a configuration file has been successfully - loaded and has changes. Bug found while making API documentation - for 1.6.0. ........ - -2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr - 2008) | 22 lines Merged revisions 114848 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr - 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel - variables instead of the channel's macrocontext and macroexten - fields. This is needed because if macros are daisy-chained, the - incorrect context and extension are placed on the new channel. I - also added locking to the channel prior to accessing these - variables as noted in trunk's janitor project file. (closes issue - #12549) Reported by: darren1713 Patches: - app_queue.c.macroextenpatch uploaded by darren1713 (license 116) - (with modifications from me) Tested by: putnopvut ........ - ................ - -2008-04-29 19:04 +0000 [r114846] Kevin P. Fleming <kpfleming@digium.com> - - * /: bug is not present in this branch - -2008-04-29 17:11 +0000 [r114831] Jason Parker <jparker@digium.com> - - * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114830 | qwell | 2008-04-29 12:10:55 -0500 - (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr - 2008) | 1 line Change warning message to debug, since there are - cases where 0 results is perfectly fine. ........ - ................ - -2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming <kpfleming@digium.com> - - * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500 - (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500 - (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr - 2008) | 2 lines stop script from appending source code if run - multiple times ........ ................ ................ - -2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 | - mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix - deadlock issue in chan_zap with libss7 due to channel variables - being set with the channel pvt lock being held. #12512 ........ - -2008-04-28 13:44 +0000 [r114714] Joshua Colp <jcolp@digium.com> - - * /, configure, configure.ac: Merged revisions 114713 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2 - lines Update autoconf logic with latest API change for libss7. - ........ - -2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 114709 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) - | 13 lines Merged revisions 114708 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) - | 5 lines When modules are embedded, they take on a different - name, without the ".so" extension. Specifically check for this - name, when we're checking if a module is loaded. (Closes issue - #12534) ........ ................ - -2008-04-27 15:20 +0000 [r114701] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6 - because it fixes a crash. ........ r114700 | mvanbaak | - 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI - in chan_skinny event based modeled after chan_zap and chan_mgcp. - (closes issue #12214) Reported by: DEA Patches: - chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested - by: DEA and me ........ - -2008-04-27 01:30 +0000 [r114697] Sean Bright <sean.bright@gmail.com> - - * /, configure, configure.ac: Merged revisions 114696 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400 - (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr - 2008) | 5 lines When we don't explicitly pass a path to the - --with-tds configure option, we may end up finding tds.h in - /usr/local/include instead of /usr/include. If this happens, the - grep that looks for the version (from tdsver.h) will fail and - we'll have some problems during the build. ........ - ................ - -2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500 - (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) - | 6 lines Clicking forward without selecting a message leaves an - errant .lock file. (closes issue #12528) Reported by: pukepail - Patches: patch.diff uploaded by pukepail (license 431) ........ - ................ - -2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 | - russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines - Lock the channel around datastore access (closes issue #12527) - Reported by: mnicholson Patches: pbx_lua4.diff uploaded by - mnicholson (license 96) ........ - - * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114674 | russell | 2008-04-25 17:00:35 -0500 - (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) - | 3 lines Use consistent logic for checking to see if a call - number has been chosen yet. Also, remove some redundant logic I - recently added in a fix. ........ ................ - -2008-04-25 19:34 +0000 [r114664] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr - 2008) | 12 lines Merged revisions 114662 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr - 2008) | 4 lines Move the unlock of the spyee channel to outside - the start_spying() function so that the channel is not unlocked - twice when using whisper mode. ........ ................ - -2008-04-25 16:26 +0000 [r114652] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr - 2008) | 4 lines Fix a memory leak and protect against potential - dereferences of a NULL pointer. ........ - -2008-04-24 22:14 +0000 [r114636] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 | - file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey - look, it builds. (closes issue #12519) Reported by: falves11 - ........ - -2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr - 2008) | 19 lines Merged revisions 114632 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr - 2008) | 11 lines Re-invite RTP during a masquerade so that, for - instance, an AMI redirect of two channels which are natively - bridged will preserve audio on both channels. This prevents a - problem with Asterisk not re-inviting due to one of the channels - having being a zombie. (closes issue #12513) Reported by: - mneuhauser Patches: - asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by - mneuhauser (license 425) ........ ................ - - * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr - 2008) | 16 lines Merged revisions 114628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr - 2008) | 8 lines Output of channel variables when - eventwhencalled=vars was set was being truncated two characters. - This patch corrects the problem. (closes issue #12493) Reported - by: davidw ........ ................ - - * channels/chan_local.c, /: Merged revisions 114625 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500 - (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr - 2008) | 10 lines Resolve a deadlock in chan_local by releasing - the channel lock temporarily. (closes issue #11712) Reported by: - callguy Patches: 11712.patch uploaded by putnopvut (license 60) - Tested by: acunningham ........ ................ - -2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 114622 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500 - (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) - | 4 lines Ensure that when we set the accountcode, it actually - shows up in the CDR. (Fix for AMI Originate) (Closes issue - #12007) ........ ................ - - * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 | - tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines - Fix DST calculation, and fix bug in calculation of whether conf - has started yet or not (Closes issue #12292) Reported by: DEA - Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license - 3) ........ - -2008-04-24 16:48 +0000 [r114613] Jason Parker <jparker@digium.com> - - * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114612 | qwell | 2008-04-24 11:47:01 -0500 - (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #12496) Reported by: daniele Patches: - misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) - Tested by: daniele Technically, I didn't use the patch above - except to find out what revision to merge - but it's the same - thing as this revision. ........ r51989 | crichter | 2007-01-24 - 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 - ........ ................ - -2008-04-24 15:57 +0000 [r114610] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114609 | russell | 2008-04-24 10:56:55 -0500 - (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) - | 4 lines Fix a silly mistake in a change I made yesterday that - caused chan_iax2 to blow up very quickly. (issue #12515) ........ - ................ - -2008-04-24 15:00 +0000 [r114607] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Merged revisions 114606 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 - lines Merged revisions 114603 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 - lines Only have one max-forwards header in outbound REFERs. - Discovered in the Asterisk SIP Masterclass in Orlando. Thanks - Joe! ........ ................ - -2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 | - russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines - Change a verbose message to debug. (closes issue #12514) ........ - - * /, main/http.c: Merged revisions 114601 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008) - | 14 lines Merged revisions 114600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) - | 6 lines Improve some broken cookie parsing code. Previously, - manager login over HTTP would only work if the mansession_id - cookie was first. Now, the code builds a list of all of the - cookies in the Cookie header. This fixes a problem observed by - users of the Asterisk GUI. (closes AST-20) ........ - ................ - - * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008) - | 18 lines Merged revisions 114597 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) - | 10 lines Fix an issue that caused getting the correct next - channel to not always work. Also, remove setting the amount of - time to wait for a digit from 5 seconds back down to 1/10 of a - second. I believe this was so the beep didn't get played over and - over really fast, but a while back I put in another fix for that - issue. (closes issue #12498) Reported by: jsmith Patches: - app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license - 15) ........ ................ - -2008-04-23 18:34 +0000 [r114596] Jason Parker <jparker@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114595 | qwell | 2008-04-23 13:33:28 -0500 - (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) | - 8 lines Fix reload/unload for res_musiconhold module. (closes - issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff - uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded - by jamesgolovich (license 176) ........ ................ - -2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant <russell@digium.com> - - * main/manager.c, /, include/asterisk/manager.h: Merged revisions - 114592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008) - | 13 lines Merged revisions 114591 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) - | 5 lines Store the manager session ID explicitly as 4 byte ID - instead of a ulong. The mansession_id cookie is coded to be - limited to 8 characters of hex, and this could break logins from - 64-bit machines in some cases. (inspired by AST-20) ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114588 | russell | 2008-04-23 12:18:29 -0500 - (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) - | 2 lines Fix find_callno_locked() to actually return the callno - locked in some more cases. ........ ................ - -2008-04-23 16:57 +0000 [r114586] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Merged revisions 114585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 - lines Merged revisions 114584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 - lines Add 502 support for both directions, not only one... (see - r114571) ........ ................ - -2008-04-23 14:56 +0000 [r114581] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, /: Merged revisions 114580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) | - 12 lines Merged revisions 114579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 - lines Instead of stopping dialplan execution when SayNumber - attempts to say a large number that it can not print out a - message informing the user and continue on. (closes issue #12502) - Reported by: bcnit ........ ................ - -2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr - 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail - changes This makes IMAP_STORAGE include the proper headers if you - have specified the "system" option for --with-imap when running - the configure script and your IMAP-related headers exist in - /usr/include/c-client. This change is due to a hasty merge of a - 1.4 change I made. ........ - -2008-04-22 23:59 +0000 [r114573] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) - | 10 lines Merged revisions 114571 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) - | 2 lines Treat a 502 just like a 503, when it comes to - processing a response code ........ ................ - -2008-04-22 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta8 released. - -2008-04-22 22:18 +0000 [r114560] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114559 | russell | 2008-04-22 17:17:31 -0500 - (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) - | 5 lines When we receive a full frame that is supposed to - contain our call number, ensure that it has the correct one. - (closes issue #10078) (AST-2008-006) ........ ................ - -2008-04-22 22:04 +0000 [r114556] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 114553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 | - murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines - (closes issue #12469) Reported by: triccyx I had a bit a problem - reproducing this in my setup (trying not to disturb my other - stuff) but finally, I got it. The problem appears to be that the - extension is being added in replace mode, which kinda assumes - that the pattern trie has been formed, when in fact, in this - case, it was not. The checks being done are not nec. when the - tree is not yet formed, as changes like this will be summarized - when the trie is formed in the future. I tested the fix, and the - crash no longer happens. Feel free to open the bug again if this - fix doesn't cure the problem. ........ - -2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 114548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 | - russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines - re-add a fix that got lost with a recent change ........ - -2008-04-22 18:14 +0000 [r114541] Jason Parker <jparker@digium.com> - - * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged - revisions 114540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 | - qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines - Allow setqueuevar=yes (et al) to work, after changes to - pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit - Patches: 12490-queuevars-3.diff uploaded by qwell (license 4) - Tested by: qwell ........ - -2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114538 | russell | 2008-04-22 13:04:39 -0500 - (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) - | 9 lines If the dial string passed to the call channel callback - does not indicate an extension, then consider the extension on - the channel before falling back to the default. (closes issue - #12479) Reported by: darren1713 Patches: - exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license - 116) ........ ................ - -2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 114527 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 | - russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines - Correct action_ping() and action_events() with regards to Manager - 1.1 documentation. Also, fix a bug in xml_translate(). (closes - issue #11649) Reported by: ys Patches: trunk_manager.c.diff - uploaded by ys (license 281) ........ - -2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 | - mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add - support for generic name transmission (#12484) on SS7 in chan_zap - ........ - -2008-04-21 15:38 +0000 [r114328] Jeff Peeler <jpeeler@digium.com> - - * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008) - | 2 lines This removes an invalid warning message for an - incorrectly entered pin, but more importantly removes an - inapplicable check. If the first argument passed to - app_authenticate does not contain a '/', the argument should be - treated as the sole fixed "password" to match against and that is - all. (Previous behavior was attempting to open a file based on - the pin.) ........ - -2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | - 12 lines Merged revisions 114322 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 - lines Only drop audio if we receive it without a progress - indication. We allow other frames through such as DTMF because - they may be needed to complete the call. (closes issue #12440) - Reported by: aragon ........ ................ - - * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6 - lines Only print out the error message if ldap_modify_ext_s - actually returns an error, and not success. (closes issue #12438) - Reported by: gservat Patches: res_config_ldap.c-patch-code - uploaded by gservat (license 466) ........ - -2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport - / Redirecting Number handling. #12425 - -2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 114285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008) - | 10 lines Merged revisions 114284 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) - | 2 lines Don't destroy a manager session if poll() returns an - error of EAGAIN. ........ ................ - - * Makefile, /: Merged revisions 114279 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008) - | 10 lines Merged revisions 114278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) - | 2 lines ensure directories are created before we try to install - stuff into them ........ ................ - - * Makefile, /: Merged revisions 114276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008) - | 10 lines Merged revisions 114275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) - | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for - bininstall ........ ................ - -2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp <jcolp@digium.com> - - * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 - lines Make sure ADSI is marked as unavailable on Unistim channels - so voicemail does not try to do some ADSI jazz. (closes issue - #12460) Reported by: PerryB ........ - -2008-04-18 18:04 +0000 [r114260] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500 - (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr - 2008) | 6 lines Clearing up error messages so they make a bit - more sense. Also removing a redundant error message. Issue AST-15 - ........ ................ - -2008-04-18 16:12 +0000 [r114255] Joshua Colp <jcolp@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4 - lines If the parsing of the config file fails make sure we unlock - ldap_lock. (closes issue #12477) Reported by: IgorG ........ - -2008-04-18 13:40 +0000 [r114247] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c: Merged revisions 114246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr - 2008) | 9 lines Merged revisions 114245 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr - 2008) | 1 line Only complete the SIP channel name once for 'sip - show channel <channel>' ........ ................ - -2008-04-18 06:54 +0000 [r114244] Tilghman Lesher <tlesher@digium.com> - - * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500 - (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008) - | 3 lines For consistency sake, ensure that the values that - ${CALLINGPRES} returns are valid as an input to SetCallingPres. - (Closes issue #12472) ........ ................ - -2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 - lines Merged revisions 114148 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 - lines Handle subscribe queues in all situations... Thanks to - festr_ on irc for telling me about this bug. ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 | - oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines - Adding chanvar to SIPPEER from 1.4 branch ........ - - * main/autoservice.c, /: Merged revisions 114233 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008) - | 14 lines Merged revisions 114230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) - | 6 lines Remove redundant safety net. The check for the - autoservice channel list state accomplishes the same goal in a - better way. (issue #12470) Reported By: atis ........ - ................ - -2008-04-17 21:05 +0000 [r114228] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr - 2008) | 17 lines Merged revisions 114226 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr - 2008) | 9 lines Declaration of the peer channel in this scope was - making it so the peer variable defined in the outer scope was - never set properly, therefore making iterating through the - channel list always restart from the beginning. This bug would - have affected anyone who called chanspy without specifying a - first argument. (closes issue #12461) Reported by: stever28 - ........ ................ - -2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson <mmichelson@digium.com> - - * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h, - include/asterisk/frame.h: Merged revisions 114208 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500 - (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr - 2008) | 12 lines It was possible for a reference to a frame which - was part of a freed DSP to still be referenced, leading to memory - corruption and eventual crashes. This code change ensures that - the dsp is freed when we are finished with the frame. This change - is very similar to a change Russell made with translators back a - month or so ago. (closes issue #11999) Reported by: destiny6628 - Patches: 11999.patch uploaded by putnopvut (license 60) Tested - by: destiny6628, victoryure ........ ................ - -2008-04-17 16:26 +0000 [r114206] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 114205 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008) - | 11 lines Merged revisions 114204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) - | 3 lines Fix the bininstall target to install from subdirs, as - well. (closes issue AST-8, patch from bmd at switchvox) ........ - ................ - -2008-04-17 15:17 +0000 [r114203] Tilghman Lesher <tlesher@digium.com> - - * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) - | 2 lines fileio.h does not exist; io.h does, though. ........ - -2008-04-17 13:55 +0000 [r114200] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008) - | 10 lines Merged revisions 114198 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) - | 2 lines Use keepalives effectively in order diagnose bug - #12432. ........ ................ - -2008-04-17 12:59 +0000 [r114197] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 114196 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008) - | 16 lines Merged revisions 114195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) - | 8 lines Add special case for when the agi cannot be executed, - to comply with the documentation that we return failure in that - case. (closes issue #12462) Reported by: fmueller Patches: - 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14) - Tested by: fmueller ........ ................ - -2008-04-17 10:56 +0000 [r114193] Sean Bright <sean.bright@gmail.com> - - * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr - 2008) | 9 lines Merged revisions 114191 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr - 2008) | 1 line Make sure we have enough room for the recording's - filename. ........ ................ - -2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr - 2008) | 14 lines Merged revisions 114184 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr - 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by - initializing the structure to all zeroes in case it contains - fields that we don't write values into (which it does as of - Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian - ........ ................ - -2008-04-15 20:53 +0000 [r114153] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 | - tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines - Oops, buffer wasn't long enough for query ........ - -2008-04-15 20:09 +0000 [r114147] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 114146 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 | - murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines - These changes: a. fix a self-found problem with SPAWN-ing an - extension, where matches were not being found b. correct some - wording in a comment c. Add some debug for future debugging. - ........ - -2008-04-15 17:22 +0000 [r114132-114142] Jason Parker <jparker@digium.com> - - * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | - 8 lines Shorten the mac address pattern, since some phones use - different identifiers (such as the i2050 softphone). (closes - issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff - uploaded by c (license 460) Tested by: c_hans ........ - - * contrib/scripts/autosupport, /: Merged revisions 114139 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114139 | qwell | 2008-04-15 12:17:37 -0500 - (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | - 7 lines Update Digium autosupport script, for more useful - information. (closes issue #12452) Reported by: angler Patches: - autosupport.diff uploaded by angler (license 106) ........ - ................ - - * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) | - 16 lines Merged revisions 114133 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | - 8 lines Allow autofill to work in the general section of - queues.conf. Additionally, don't try to (re)set options when they - have empty values in realtime (all unset columns would have an - empty value). (closes issue #12445) Reported by: atis Patches: - 12445-autofill.diff uploaded by qwell (license 4) ........ - ................ - -2008-04-14 18:34 +0000 [r114122] Jason Parker <jparker@digium.com> - - * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114121 | qwell | 2008-04-14 13:34:17 -0500 - (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | - 7 lines The call_token on the pvt can occasionally be NULL, - causing a crash. If it is NULL, we can skip this channel, since - it can't the one we're looking for. (closes issue #9299) Reported - by: vazir ........ ................ - -2008-04-14 17:42 +0000 [r114119] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 114118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr - 2008) | 19 lines Merged revisions 114117 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr - 2008) | 11 lines Increase the retry count when attempting to show - channels. This apparently cleared an issue someone was seeing - when attempting to show channels when the load was high. (closes - issue #11667) Reported by: falves11 Patches: 11677.txt uploaded - by russell (license 2) Tested by: falves11 ........ - ................ - -2008-04-14 16:33 +0000 [r114116] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008) - | 2 lines Make tab-completion work for all cases ........ - -2008-04-14 16:25 +0000 [r114114] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500 - (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr - 2008) | 9 lines If the datastore has been moved to another - channel due to a masquerade, then freeing the datastore here - causes an eventual double free when the new channel hangs up. We - should only free the datastore if we were able to successfully - remove it from the channel we are referencing (i.e. the datastore - was not moved). (closes issue #12359) Reported by: pguido - ........ ................ - -2008-04-14 15:02 +0000 [r114108] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 114107 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr - 2008) | 13 lines Merged revisions 114106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr - 2008) | 5 lines Save a local copy of the generate callback prior - to unlocking the channel in case the generate callback goes NULL - on us after the channel is unlocked. Thanks to Russell for - pointing this need out to me. ........ ................ - -2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | - 12 lines Merged revisions 114103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 - lines It is possible for the remote side to say they want T38 but - not give any capabilities. (closes issue #12414) Reported by: MVF - ........ ................ - - * main/rtp.c, /: Merged revisions 114101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | - 12 lines Merged revisions 114100 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 - lines Don't change the SSRC when a new source comes into play, - this might happen quite often and depending on the remote side... - they might not like this. (closes issue #12353) Reported by: - dimas ........ ................ - -2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008) - | 3 lines Add tab command-line completion (Closes issue #12428) - ........ - - * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 | - tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines - Use ast_mkdir instead of mkdir (Closes issue #12430) ........ - -2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure linkset is locked exiting - ss7_start_call - - * channels/chan_zap.c: Make sure we start incoming calls on SS7 - with echo cancellation enabled. Also make sure when completing a - COT we call ss7_start_call with the proper locks held. Lastly, - make sure if we fail to get a channel from zt_new that we don't - assume it's there. - -2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 | - tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines - If any field is not null, but has no default, then it must be set - or the insert will fail. (Closes issue #12285) ........ - - * /, configs/res_ldap.conf.sample: Merged revisions 114088 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 - Apr 2008) | 3 lines Make the sample config match the contributed - LDAP schema (Closes issue #12421) ........ - -2008-04-11 23:21 +0000 [r114087] Terry Wilson <twilson@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r114084 | twilson | 2008-04-11 17:48:52 -0500 - (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) - | 7 lines Several places in the code called find_callno() (which - releases the lock on the pvt structure) and then immediately - locked the call and did things with it. Unfortunately, the call - can disappear between the find_callno and the lock, causing Bad - Stuff(tm) to happen. Added find_callno_locked() function to - return the callno withtout unlocking for instances that it is - needed. (issue #12400) Reported by: ztel ........ - ................ - -2008-04-11 23:13 +0000 [r114086] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008) - | 7 lines Use the correct function for free'ing objects, and - maybe we won't crash. (closes issue #12163) Reported by: gservat - Patches: 20080411__bug12163.diff.txt uploaded by Corydon76 - (license 14) Tested by: gservat ........ - -2008-04-11 15:51 +0000 [r114065] Mark Michelson <mmichelson@digium.com> - - * /, main/features.c: Merged revisions 114064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr - 2008) | 19 lines Merged revisions 114063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr - 2008) | 11 lines Fix a race condition that may happen between a - sip hangup and a "core show channel" command. This patch adds - locking to prevent the resulting crash. (closes issue #12155) - Reported by: tsearle Patches: show_channels_crash2.patch uploaded - by tsearle (license 373) Tested by: tsearle ........ - ................ - -2008-04-11 14:56 +0000 [r114062] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008) - | 6 lines Errors are all greater than 0 (closes issue #12422) - Reported by: nito Patches: - res_config_ldap_result_check_patch.diff uploaded by nito (license - 340) ........ - -2008-04-10 22:23 +0000 [r114056] Mark Michelson <mmichelson@digium.com> - - * utils/conf2ael.c, utils/check_expr.c, utils/Makefile, - main/manager.c, /, utils/astman.c, utils/hashtest.c, - main/utils.c, include/asterisk/lock.h, utils/ael_main.c, - utils/hashtest2.c: Merged revisions 114052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr - 2008) | 11 lines Merged revisions 114051 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr - 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. - ........ ................ - -2008-04-10 19:59 +0000 [r114047] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr - 2008) | 14 lines Merged revisions 114045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr - 2008) | 6 lines Be sure that we're not about to set bridgepvt - NULL prior to dereferencing it. (closes issue #11775) Reported - by: fujin ........ ................ - -2008-04-10 19:09 +0000 [r114043] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008) - | 7 lines The hydra grows yet another head... (closes issue - #12401) Reported by: davevg Patches: astcli.diff2 uploaded by - davevg (license 209) Tested by: davevg, Corydon76 ........ - -2008-04-10 17:27 +0000 [r114037] Jason Parker <jparker@digium.com> - - * /, main/file.c: Merged revisions 114036 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) | - 18 lines Merged revisions 114035 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | - 10 lines Only try to prefix language if we are not using an - absolute path (suffix it otherwise). - en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes - issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff - uploaded by qwell (license 4) Tested by: kuj, qwell ........ - ................ - -2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | - 14 lines Merged revisions 114029 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 - lines Create the directory where name recordings will go if it - does not exist. (closes issue #12311) Reported by: rkeene - Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ - ................ - - * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6 - lines Don't hardcode ru into the digits filename so that - languageprefix can work. (closes issue #12404) Reported by: IgorG - Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG - (license 20) ........ - - * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c: - Merged revisions 114024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 | - file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix - spelling of existent in a few places. (closes issue #12409) - Reported by: candlerb ........ - - * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | - 14 lines Merged revisions 114021 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 - lines Don't add custom URI options if they don't exist OR they - are empty. (closes issue #12407) Reported by: homesick Patches: - uri_options-1.4.diff uploaded by homesick (license 91) ........ - ................ - -2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 | - mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8 - lines Fix a crash that happened due to accessing free'd memory - (closes issue #12396) Reported by: tcalosi Patches: 12396.patch - uploaded by putnopvut (license 60) Tested by: tcalosi ........ - - * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr - 2008) | 16 lines Merged revisions 113927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr - 2008) | 8 lines We need to set the persistant_route [sic] - parameter for the sip_pvt during the initial INVITE, no matter if - we're building the route set from an INVITE request or response. - (closes issue #12391) Reported by: benjaminbohlmann Tested by: - benjaminbohlmann ........ ................ - -2008-04-09 19:02 +0000 [r113876] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions - 113875 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) - | 12 lines Merged revisions 113874 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) - | 4 lines If the [csv] section does not exist in cdr.conf, then - an unload/load sequence is needed to correct the problem. Track - whether the load succeeded with a variable, so we can fix this - with a simple reload event, instead. ........ ................ - -2008-04-09 17:56 +0000 [r113839] Jason Parker <jparker@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) | - 2 lines Fix a small file handle "leak" pointed out by jjshoe on - #asterisk. ........ - -2008-04-09 17:50 +0000 [r113837] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, /: Merged revisions 113836 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 | - mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 - lines There was a subtle logical difference between 1.4 and trunk - with regards to how timeouts were handled. In 1.4, if the - absolute timeout were reached on a call, no matter what the - return value of ast_spawn_extension was, the pbx would attempt to - go to the 'T' extension or hangup otherwise. The rearrangement of - this function in trunk made this check only happen in the case - that ast_spawn_extension returned 0. If ast_spawn_extension - returned 1, then the fact that the timeout expired resulted in a - no-op, and would cause an infinite loop to occur in - __ast_pbx_run. This change fixes this problem. Now timeouts will - behave as they did in 1.4 (closes issue #11550) Reported by: pj - Tested by: putnopvut ........ - -2008-04-09 16:53 +0000 [r113786] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, - 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 - lines If we receive an AUTHREQ from the remote server and we are - unable to reply (for example they have a secret configured, but - we do not) then queue a hangup frame on the Asterisk channel. - This will cause the channel to hangup and a HANGUP to be sent via - IAX2 to the remote side which is the proper thing to do in this - scenario. (closes issue #12385) Reported by: viraptor ........ - ................ - -2008-04-09 14:42 +0000 [r113683] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr - 2008) | 17 lines Merged revisions 113681 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr - 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a - reinvite), then we should not send a BYE. (closes issue #12392) - Reported by: fnordian Patches: chan_sip.patch uploaded by - fnordian (license 110) with small modification from me ........ - ................ - -2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008) - | 6 lines Additional enhancements (closes issue #12390) Reported - by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir - (license 46) ........ - -2008-04-09 01:40 +0000 [r113598] Terry Wilson <twilson@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113597 | twilson | 2008-04-08 20:36:58 -0500 - (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) - | 2 lines Initialize fr->cacheable to make valgrind happy - ........ ................ - -2008-04-08 21:34 +0000 [r113560] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astcli (added): Merged revisions 113559 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08 - Apr 2008) | 6 lines Add commandline tool for doing CLI commands - through AMI (instead of using asterisk -rx) (closes issue #12389) - Reported by: davevg Patches: astcli uploaded by davevg (license - 209) ........ - -2008-04-08 18:49 +0000 [r113404-113506] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113505 | qwell | 2008-04-08 13:49:21 -0500 - (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr - 2008) | 1 line Add a little more that is required for previously - added devices. ........ ................ - - * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113455 | qwell | 2008-04-08 13:08:35 -0500 - (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | - 4 lines Add support for several new(ish) devices - most notably, - 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing - me the required information. ........ ................ - - * main/asterisk.c, /: Merged revisions 113403 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) | - 9 lines Merged revisions 113402 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | - 1 line Work around some silliness caused by sys/capability.h - - this should fix compile errors a number of users have been - experiencing. ........ ................ - -2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500 - (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) - | 6 lines Add security note on astgenkey's manpage. (closes issue - #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt - uploaded by Corydon76 (license 14) ........ ................ - - * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008) - | 15 lines Merged revisions 113348 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) - | 7 lines Move check for still-bridged channels out a little - further, to avoid possible deadlocks. (Closes issue #12252) - Reported by: callguy Patches: 20080319__bug12252.diff.txt - uploaded by Corydon76 (license 14) Tested by: callguy ........ - ................ - -2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp <jcolp@digium.com> - - * /, main/audiohook.c: Merged revisions 113297 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | - 12 lines Merged revisions 113296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 - lines If audio suddenly gets fed into one side of a channel after - a lapse of frames flush the other factory so that old audio does - not remain in the factory causing the sync code to not execute. - (closes issue #12296) Reported by: jvandal ........ - ................ - -2008-04-07 22:17 +0000 [r113246] Tilghman Lesher <tlesher@digium.com> - - * /, configs/manager.conf.sample: Merged revisions 113245 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 - Apr 2008) | 2 lines Additional note ........ - -2008-04-07 21:49 +0000 [r113244] Jason Parker <jparker@digium.com> - - * /, configs/manager.conf.sample: Merged revisions 113243 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr - 2008) | 1 line Document 'originate' permission in manager sample - config. ........ - -2008-04-07 21:36 +0000 [r113242] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) - | 23 lines Merged revisions 113013 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) - | 15 lines Merged revisions 113012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) - | 7 lines (closes issue #12362) (closes issue #12372) Reported - by: vinsik Tested by: tecnoxarxa This one line change makes an if - inside a for loop (in realtime_peer) check all the ast_variables - the loop was intending to test rather than just the first one. - ........ ................ ................ - -2008-04-07 19:10 +0000 [r113174] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged - revisions 113119 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | - 16 lines Merged revisions 113118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | - 8 lines Allow playback with noanswer (and add earlyrtp option). - (closes issue #9077) Reported by: pj Patches: earlyrtp.diff - uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, - wedhorn ........ ................ - -2008-04-07 19:08 +0000 [r113173] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500 - (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) - | 3 lines Force ast_mktime() to check for DST, since strptime(3) - does not. (Closes issue #12374) ........ ................ - -2008-04-07 16:13 +0000 [r113067] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 113066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr - 2008) | 21 lines Merged revisions 113065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr - 2008) | 13 lines This fix prevents a deadlock that was - experienced in chan_local. There was deadlock prevention in place - in chan_local, but it would not work in a specific case because - the channel was recursively locked. By unlocking the channel - prior to calling the generator's generate callback in - ast_read_generator_actions(), we prevent the recursive locking, - and therefore the deadlock. (closes issue #12307) Reported by: - callguy Patches: 12307.patch uploaded by putnopvut (license 60) - Tested by: callguy ........ ................ - -2008-04-07 15:28 +0000 [r113042] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) - | 15 lines Merged revisions 113012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) - | 7 lines (closes issue #12362) (closes issue #12372) Reported - by: vinsik Tested by: tecnoxarxa This one line change makes an if - inside a for loop (in realtime_peer) check all the ast_variables - the loop was intending to test rather than just the first one. - ........ ................ - -2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 112972 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 | - tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines - AsyncAGI should not close the manager session on error. (closes - issue #12370) Reported by: srt Patches: asterisk-12370.diff - uploaded by srt (license 378) ........ - -2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200 - (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 - Apr 2008) | 1 line Free newly allocated channel before returning - ........ ................ - - * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200 - (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) - | 7 lines Prevent call connections when codecs don't match. - (closes issue #10604) Reported by: keepitcool Patches: - branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested - by: phsultan ........ ................ - -2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/asterisk.c, /: Merged revisions 112653,112656,112714 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03 - Apr 2008) | 1 line add a Zaptel timer check to verify the timer - is responding when Zaptel support is compiled into Asterisk and - Zaptel drivers are loaded. This will help people not waste their - valuable time debugging side effects. ........ r112656 | dhubbard - | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy - buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500 - (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel - timer error message to display before exit ........ - -2008-04-04 00:54 +0000 [r112713] Joshua Colp <jcolp@digium.com> - - * /, main/Makefile: Merged revisions 112712 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | - 10 lines Merged revisions 112711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 - lines Pass in the path to Zaptel for systems that install Zaptel - headers in a separate location. ........ ................ - -2008-04-03 14:42 +0000 [r112601] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr - 2008) | 17 lines Merged revisions 112599 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr - 2008) | 9 lines Fix the testing of the "res" variable so that it - is more logically correct and makes the correct warning and debug - messages print. (closes issue #12361) Reported by: one47 Patches: - chan_zap_deferred_digit.patch uploaded by one47 (license 23) - ........ ................ - -2008-04-02 17:37 +0000 [r112470] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 112469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr - 2008) | 21 lines Merged revisions 112468 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr - 2008) | 13 lines Fix a race condition in the manager. It is - possible that a new manager event could be appended during a - brief time when the manager is not waiting for input. If an event - comes during this period, we need to set an indicator that there - is an event pending so that the manager doesn't attempt to wait - forever for an event that already happened. (closes issue #12354) - Reported by: bamby Patches: manager_race_condition.diff uploaded - by bamby (license 430) (comments added by me) ........ - ................ - -2008-04-02 15:27 +0000 [r112436] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 | - file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines - Since the SIP request structure gets reused multiple times with - TCP handling we have to clear the debug state or else we will - keep spitting out debug even after it has been turned off. - (closes issue #12169) Reported by: pj Patches: - 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj - ........ - -2008-04-02 14:33 +0000 [r112395] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr - 2008) | 14 lines Merged revisions 112393 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr - 2008) | 6 lines Ensure that there is no timeout if none is - specified. (closes issue #12349) Reported by: johnlange ........ - ................ - -2008-04-01 22:48 +0000 [r112359] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 112357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 | - murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line - Bumped across another test set for the new exten pattern matcher, - which revealed a problem with the CANMATCH/MATCHMORE modes. - Direct matches were getting in the way. Fixed. ........ - -2008-04-01 20:20 +0000 [r112299] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 112289 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 | - murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines - (closes issue #12298) Reported by: falves11 Patches: 12298.patch1 - uploaded by murf (license 17) Tested by: murf I have hopes that - the changes made over the last few days will finalize and - solidify this code. While there are bound to be small tweaks - still needed, I feel that the job (at last) is somewhat - completed. Finally, I had a chance to comprehend how the scoring - of extension patterns was done in the previous version, and I've - come very close to using the exact same criteria in the new - pattern matching code. The left-right sorting is now replicated - in the trie structure itself, such that the first match found - will the 'best' match. Compared the results against 1.4 for - several extensions. Replicated falves11's setup and it works. - Used some devious patterns provided by jsmith, supplemented with - a few of my own. Looks good. ........ - -2008-04-01 18:09 +0000 [r112211] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 112210 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | - 12 lines Merged revisions 112209 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 - lines Disable Packet2Packet bridging when we need to feed DTMF - frames into the core. Some implementations do not like how we - switch between things. (closes issue #12212) Reported by: bamby - ........ ................ - -2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | - 12 lines Merged revisions 112204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 - lines Do not pass audio until the remote side has indicated they - are providing early media, or if the channel has been answered. - (closes issue #11823) Reported by: SDamm ........ - ................ - -2008-04-01 17:25 +0000 [r112157] Mark Michelson <mmichelson@digium.com> - - * main/dns.c, /: Merged revisions 112148 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr - 2008) | 18 lines Merged revisions 112138 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr - 2008) | 10 lines Initialize the __res_state structure used for - dns purposes to all 0's prior to using it. This is due to - valgrind's complaints on issue #12284 as well as an excerpt found - in "Description" portion of the online man page found here: - http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV - (pertains to issue #12284 but does not necessarily close it) - ........ ................ - -2008-04-01 16:57 +0000 [r112127] Joshua Colp <jcolp@digium.com> - - * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged - revisions 112126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) | - 13 lines Merged revisions 112125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 - lines Ensure that we do not exceed the hold's maximum size with a - single frame. (closes issue #12047) Reported by: fabianoheringer - Tested by: fabianoheringer ........ ................ - -2008-03-31 22:17 +0000 [r112070-112072] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r112069 | qwell | 2008-03-31 16:48:30 -0500 - (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | - 5 lines Fix a silly infinite loop when choosing an invalid - option. (closes issue #12315) Reported by: jmls ........ - ................ - -2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson <twilson@digium.com> - - * /, main/http.c: Merged revisions 112033 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 | - twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines - Handle blank prefix= in http.conf ........ - -2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 111998 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 | - russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines - Ensure configure gets run on a clean checkout. (closes issue - #12197) Reported by: juggie Patches: 12197.diff uploaded by - juggie (license 24) ........ - -2008-03-31 14:22 +0000 [r111962] Joshua Colp <jcolp@digium.com> - - * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4 - lines Initialize all these here tmp pointers at declaration. They - confused some compilers a wee bit. (closes issue #12333) Reported - by: ovi ........ - -2008-03-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta7.1 released. - - Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch. - -2008-03-28 21:46 +0000 [r111858] Jason Parker <jparker@digium.com> - - * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111857 | qwell | 2008-03-28 16:46:02 -0500 - (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | - 12 lines Allow gsm to compile correctly on x86 with gcc4 - optimizations. (closes issue #11243) Reported by: whiskerp - Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) - Tested by: Seggy (IRC) Note: While I did write this patch, I - would not have found this if fossil had not reported and fixed - issue #12253. A huge thanks to him for helping to (indirectly) - find the problem here. ........ ................ - -2008-03-28 19:11 +0000 [r111722-111776] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111721 | qwell | 2008-03-28 12:57:12 -0500 - (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar - 2008) | 1 line Remove unimplemented softkeys. Prompted by issue - #12325. ........ ................ - -2008-03-28 16:21 +0000 [r111660] Jason Parker <jparker@digium.com> - - * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111659 | qwell | 2008-03-28 11:20:59 -0500 - (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | - 8 lines The file size of WAV49 does not need to be an even - number. (closes issue #12128) Reported by: mdu113 Patches: - 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: - qwell, mdu113 ........ ................ - -2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher <tlesher@digium.com> - - * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) - | 11 lines Merged revisions 111605 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) - | 3 lines Update debugging text, since Valgrind eliminated the - --log-file-exactly option. (Closes issue #12320) ........ - ................ - -2008-03-28 00:56 +0000 [r111566] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 | - file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines - Forgetting to unregister a manager action is bad, mmmk? ........ - -2008-03-28 00:17 +0000 [r111534] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 | - mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10 - lines Fix a crash that would happen when attempting to unload the - app_queue module. The problem was that when the refcount on the - queue hit 0, the destructor was called, and inside the - destructor, another function was called which would increase the - refcount back to 1 again and then decrease it again back to 0 for - every member in the queue. This meant that the destructor was - being recursively called, leading to a double free of the queue. - This is now fixed by making sure to unlink the queue from the - queues container prior to the final unref of the queue. ........ - -2008-03-27 21:28 +0000 [r111498] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 111497 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 | - murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line - comment cleanup and iron out a really dumb mistake in handling - the '.'-wildcard in the new exten pattern matcher. ........ - -2008-03-27 19:30 +0000 [r111444] Tilghman Lesher <tlesher@digium.com> - - * /, main/acl.c: Merged revisions 111443 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) - | 14 lines Merged revisions 111442 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) - | 6 lines For FreeBSD, at least, the ifa_addr element could be - NULL. (closes issue #12300) Reported by: festr Patches: - acl.c.patch uploaded by festr (license 443) ........ - ................ - -2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy <murf@digium.com> - - * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, - 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 - lines These small documentation updates made in response to a - query in asterisk-users, where a user was using Playback, but - needed the features of Background, and had no idea that - Background existed, or that it might provide the features he - needed. I thought the best way to avert these kinds of queries - was to provide "See Also" references in all three of - "Background", "Playback", "WaitExten". Perhaps a project to do - this with all related apps is in order. ........ ................ - - * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c, - include/asterisk/ael_structs.h: Merged revisions 111360 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, - 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | - 15 lines (closes issue #12302) Reported by: pj Tested by: murf - These changes will set a channel variable ~~EXTEN~~ just before - generating code for a switch, with the value of ${EXTEN}. The - exten is marked as having a switch, and ever after that, till the - end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} - instead in application arguments; (and the ${EXTEN: also). The - reason for this, is that because switches are coded using - separate extensions to provide pattern matching, and jumping - to/from these switch extensions messes up the ${EXTEN} value, - which blows the minds of users. ........ ................ - -2008-03-27 00:36 +0000 [r111247-111339] Jason Parker <jparker@digium.com> - - * main/frame.c, /: Merged revisions 111285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | - 9 lines Merged revisions 111280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | - 1 line Put this flag back so we don't change the API. ........ - ................ - - * main/frame.c, /: Merged revisions 111246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | - 17 lines Merged revisions 111245 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | - 9 lines Remove excessive smoother optimization that was causing - audio glitches (small "pops") after (about 200ms later) an - "incorrectly" sized frame was received. While it would be very - nice to keep this as optimized as possible, it makes no sense for - the smoother to be dropping random bits of audio like this. Isn't - that the whole point of a smoother? Closes issue #12093. ........ - ................ - -2008-03-26 19:57 +0000 [r111131] Joshua Colp <jcolp@digium.com> - - * contrib/scripts/autosupport, /: Merged revisions 111130 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed, - 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 - lines Update autosupport script. (closes issue #12310) Reported - by: angler Patches: autosupport.diff uploaded by angler (license - 106) ........ ................ - -2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming <kpfleming@digium.com> - - * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar - 2008) | 18 lines Merged revisions 111126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 - (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar - 2008) | 2 lines update UPGRADE notes to document usage of the - script ........ ................ ................ - -2008-03-26 19:41 +0000 [r111124] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500 - (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar - 2008) | 4 lines This code change is made just for clarification. - It does exactly the same thing as before. It just doesn't look as - wrong. ........ ................ - -2008-03-26 19:27 +0000 [r111072] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500 - (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar - 2008) | 9 lines Add a lock to the vm_state structure and use the - lock around mail_open calls to prevent concurrent access of the - same mailstream. This, along with trunk's ability to configure - TCP timeouts for IMAP storage will help to prevent crashes and - hangs when using voicemail with IMAP storage. (closes issue - #10487) Reported by: ewilhelmsen ........ ................ - -2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): - Merged revisions 111025 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar - 2008) | 18 lines Merged revisions 111024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500 - (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar - 2008) | 2 lines add a script to make getting the iLBC source code - simple for end users ........ ................ ................ - -2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) | - 12 lines Merged revisions 111020 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 - lines If we are requested to authenticate a reinvite make sure - that it contains T38 SDP if need be. (closes issue #11995) - Reported by: fall ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed, - 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 - lines Add an option (transmit_silence) which transmits silence - during both Record() and DTMF generation. The reason this is an - option is that in order to transmit silence we have to setup a - translation path. This may not be needed/wanted in all cases. - (closes issue #10058) Reported by: tracinet ........ - ................ - -2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming <kpfleming@digium.com> - - * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar - 2008) | 10 lines Merged revisions 110962 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar - 2008) | 2 lines add note that the user will need to enable - codec_ilbc to get it to build ........ ................ - -2008-03-26 17:35 +0000 [r110959] Donny Kavanagh <donnyk@gmail.com> - - * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 | - juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines - update documentation to reflect the changes in the way configure - detects net-snmp. (closes issue #12067) Reported by: juggie - Patches: 12067_snmp_doc.patch uploaded by juggie (license 24) - Tested by: juggie ........ - -2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h - (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile, - codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), - codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c - (removed), codecs/ilbc/iCBSearch.h (removed), - codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), - codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c - (removed), codecs/ilbc/hpOutput.h (removed), - codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, - codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c - (removed), codecs/ilbc/StateSearchW.h (removed), - codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h - (removed), codecs/ilbc/syntFilter.h (removed), - codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c - (removed), codecs/ilbc/packing.h (removed), - codecs/ilbc/libilbc.vcproj (removed), - codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c - (removed), codecs/ilbc/getCBvec.c (removed), - codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed), - codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h - (removed), codecs/ilbc/LPCdecode.h (removed), - codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c - (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h - (removed), codecs/ilbc/iLBC_encode.h (removed), - codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c - (removed), codecs/ilbc/doCPLC.c (removed), - codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h - (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h - (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, - codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c - (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged - revisions 110881 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar - 2008) | 18 lines Merged revisions 110880 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 - (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar - 2008) | 2 lines due to licensing restrictions, we cannot - distribute the source code for iLBC encoding and decoding... so - remove it, and add instructions on how the user can obtain it - themselves ........ ................ ................ - -2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp <jcolp@digium.com> - - * /: Merged revisions 110726 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 | - jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines - This one line change makes an if inside a for loop (in - realtime_peer) check all the ast_variables the loop was intending - to test rather than just the first one. ........ - -2008-03-26 00:03 +0000 [r110832] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 110831 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 | - mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 - lines This ensures that the manager interface is not enabled by - default. Prior to this change, it was possible to start Asterisk - with the manager interface enabled, then either comment out the - enabled option or make manager.conf unopenable and the manager - interface would still be enabled. ........ - -2008-03-25 22:52 +0000 [r110781] Jason Parker <jparker@digium.com> - - * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | - 14 lines Merged revisions 110779 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | - 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes - issue #12268. Patch borrowed from r82344 ........ - ................ - -2008-03-25 22:11 +0000 [r110778] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: This one line change makes an if inside a - for loop (in realtime_peer) check all the ast_variables the loop - was intending to test rather than just the first one. - -2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher <tlesher@digium.com> - - * configs/extensions.conf.sample, /, configs/voicemail.conf.sample: - Merged revisions 110691 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 | - tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines - Update sample configurations to make virtual hosting more - obvious. (closes issue #11969) Reported by: pprindeville Patches: - acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) - ........ - - * configs/extensions.conf.sample, /: Merged revisions 110689 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 - Mar 2008) | 6 lines Update the sample configuration, to use Macro - less (since it's now deprecated). (closes issue #12293) Reported - by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by - pprindeville (license 347) ........ - -2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Oops. - - * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar - 2008) | 15 lines Merged revisions 110635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar - 2008) | 7 lines When reverting a commit, I accidentally left in - this bit which was an experiment to see what would happen. It - passed the compile test, and I didn't notice I had left this - change in too. So this is a revert of a revert...sort of. - ........ ................ - -2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp <jcolp@digium.com> - - * include/asterisk/options.h, main/asterisk.c, Makefile, /, - main/app.c: Merged revisions 110629 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) | - 12 lines Merged revisions 110628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 - lines Add an option (transmit_silence) which transmits silence - during both Record() and DTMF generation. The reason this is an - option is that in order to transmit silence we have to setup a - translation path. This may not be needed/wanted in all cases. - (closes issue #10058) Reported by: tracinet ........ - ................ - -2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar - 2008) | 23 lines Merged revisions 110618 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar - 2008) | 15 lines This is a revert for revision 108288. The reason - is that that revision was not for an actual bug fix per se, and - so it really should not have been in 1.4 in the first place. - Plus, people who compile with DO_CRASH are more likely to - encounter a crash due to this change. While I think the usage of - DO_CRASH in ast_sched_del is a bit absurd, this sort of change is - beyond the scope of 1.4 and should be done instead in a developer - branch based on trunk so that all scheduler functions are fixed - at once. I also am reverting the change to trunk and 1.6 since - they also suffer from the DO_CRASH potential. (closes issue - #12272) Reported by: qq12345 ........ ................ - -2008-03-24 17:36 +0000 [r110616] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r110615 | russell | 2008-03-24 12:36:04 -0500 - (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) - | 2 lines Turn a NOTICE into a DEBUG message. ........ - ................ - -2008-03-24 15:29 +0000 [r110611] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 | - file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines - Only print out the set_address_from_contact host verbose message - if debugging is enabled on the dialog. (closes issue #12280) - Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain - (license 226) ........ - -2008-03-21 21:52 +0000 [r110579] Jason Parker <jparker@digium.com> - - * /, sounds/Makefile: Merged revisions 110578 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 | - qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line - Update to 1.4.11 core sounds. ........ - -2008-03-21 15:25 +0000 [r110501] Russell Bryant <russell@digium.com> - - * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 - Mar 2008) | 3 lines Note that the TCP and TLS support is - currently considered experimental and is subject to change while - we work out the remaining issues. ........ - -2008-03-21 14:36 +0000 [r110476] Jason Parker <jparker@digium.com> - - * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | - 15 lines Merged revisions 110474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | - 7 lines Don't attempt to do optimizations of gsm on mips - platforms either. (closes issue #12270) Reported by: zandbelt - Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) - ........ ................ - -2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: Merged revisions 110396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) - | 17 lines Merged revisions 110395 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) - | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms - in the autoservice thread. This really should not make a - difference except in very rare cases. That case would be that all - of the channels in autoservice are not generating any frames. In - that case, this change reduces the potential amount of time that - a thread waits in ast_autoservice_stop() for the autoservice - thread to wrap back around to the beginning of its loop. (closes - issue #12266, reported by dimas) ........ ................ - - * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 | - russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines - Use the correct buffer for g722tolin16_sample. This shouldn't - have caused any problems, but Qwell noticed the typo here. - ........ - - * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions - 110337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) - | 22 lines Merged revisions 110336 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r110336 | russell | 2008-03-20 16:54:58 -0500 - (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) - | 6 lines Fix some very broken code that was introduced in 1.2.26 - as a part of the security fix. The dnsmgr is not appropriate - here. The dnsmgr takes a pointer to an address structure that a - background thread continuously updates. However, in these cases, - a stack variable was passed. That means that the dnsmgr thread - would be continuously writing to bogus memory. ........ - ................ ................ - - * /, main/file.c: Merged revisions 110303 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 | - russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines - Fix a bug when using zaptel timing for playing back files that - have a sample rate other than 8 kHz. The issue here is that - format modules give a "whennext" sample value, which is used to - calculate when to set a timer for to retrieve the next frame. - However, the zaptel timer operates on 8 kHz samples, so this must - be taken into account. (another part of issue #12164, reported by - milazzo and jsmith, patch by me) ........ - -2008-03-20 18:02 +0000 [r110273] Mark Michelson <mmichelson@digium.com> - - * main/dial.c, /: Merged revisions 110272 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 | - mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 - lines Add missing unlock ........ - -2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant <russell@digium.com> - - * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 - Mar 2008) | 27 lines Add some fixes that I made in regards to - wideband codec handling to get G.722 music on hold working for - me. (issue #12164, reported by milazzo and jsmith, patches by me) - res/res_musiconhold.c: - I moved a single line so that the sample - queue update happened before ast_write(). The reason that this - was a bug is that the G.722 frame originally says it has 320 - samples in it (which is correct). However, when the frame is - written to a channel that uses RTP, main/rtp.c modifies the frame - to cut the number of samples in half before it sends it on the - wire. This is to account for the stupid incorrect G.722 spec that - makes it so we have to lie about the number of samples with RTP. - I should probably go and re-work the RTP code so it doesn't - modify the frame so that a bug like this won't happen in the - future. However, this change to MOH is harmless. main/channel.c: - - I made two fixes in regards to generator timing. Generators use - samples for timing. However, this code assumed 8 kHz samples. In - one case, it was a hard coded 160 samples, that is now written as - the sample rate / 50. The other place was dealing with timing a - generator based on frames coming from the other direction. - However, that would have only worked if the sample rates for the - formats in both directions were the same. The code now takes into - account that the sample rates may differ, and scales the - generator samples accordingly. ........ - -2008-03-19 23:00 +0000 [r110165] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) - | 13 lines Merged revisions 110163 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) - | 5 lines Fix a bug where when calls on the trunk side hang up - while on hold, the state is not properly reflected. (closes issue - #11990, reported by anakaoka, patched by me) ........ - ................ - -2008-03-19 21:06 +0000 [r110088] Jeff Peeler <jpeeler@digium.com> - - * /: marking rev 110087 from trunk as not applying - -2008-03-19 20:37 +0000 [r110085] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar - 2008) | 12 lines Merged revisions 110083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar - 2008) | 4 lines Add a missing unlock in the case that memory - allocation fails in app_chanspy. Thanks to Russell for confirming - that this was an issue. ........ ................ - -2008-03-19 19:14 +0000 [r110037] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, - 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 - lines Add sanity checking for position resuming. We *have* to - make sure that the position does not exceed the total number of - files present, and we have to make sure that the position's - filename is the same as previous. These values can change if a - music class is reloaded and give unpredictable behavior. (closes - issue #11663) Reported by: junky ........ ................ - -2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant <russell@digium.com> - - * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml - (added), /: Merged revisions 109974 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | - 13 lines Merged revisions 109973 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | - 5 lines People report bugs about Asterisk crashing with DO_CRASH - enabled was getting a little silly... Now we only show certain - cflags when you run configure with --enable-dev-mode - (corresponding menuselect change to follow) ........ - ................ - -2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 110020 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | - 14 lines Merged revisions 110019 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 - lines Make sure that the mark bit does not incorrectly cause - video frame timestamps to be calculated as if they are audio - frames. (closes issue #11429) Reported by: sperreault Patches: - 11429-frametype.diff uploaded by qwell (license 4) ........ - ................ - -2008-03-19 16:46 +0000 [r109969] Steve Murphy <murf@digium.com> - - * main/config.c, /: Merged revisions 109942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) | - 80 lines Merged revisions 109908 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | - 72 lines (closes issue #11442) Reported by: tzafrir Patches: - 11442.patch uploaded by murf (license 17) Tested by: murf I - didn't give tzafrir very much time to test this, but if he does - still have remaining issues, he is welcome to re-open this bug, - and we'll do what is called for. I reproduced the problem, and - tested the fix, so I hope I am not jumping by just going ahead - and committing the fix. The problem was with what file_save does - with templates; firstly, it tended to print out multiple options: - [my_category](!)(templateref) instead of - [my_category](!,templateref) which is fixed by this patch. - Nextly, the code to suppress output of duplicate declarations - that would occur because the reader copies inherited declarations - down the hierarchy, was not working. Thus: [master-template](!) - mastervar = bar [template](!,master-template) tvar = value - [cat](template) catvar = val would be rewritten as: ;! ;! - Automatically generated configuration file ;! Filename: - experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: - Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! - [master-template](!) mastervar = bar - [template](!,master-template) mastervar = bar tvar = value - [cat](template) mastervar = bar tvar = value catvar = val This - has been fixed. Since the config reader 'explodes' inherited vars - into the category, users may, in certain circumstances, see - output different from what they originally entered, but it should - be both correct and equivalent. ........ ................ - -2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant <russell@digium.com> - - * /, main/utils.c: Merged revisions 109839 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) - | 10 lines Merged revisions 109838 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) - | 2 lines Tweak spacing in a recent change because I'm very - picky. ........ ................ - - * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) - | 11 lines Merged revisions 109763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) - | 3 lines Fix one place where the chanspy datastore isn't removed - from a channel. (issue #12243, reported by atis, patch by me) - ........ ................ - -2008-03-18 23:23 +0000 [r109779] Tilghman Lesher <tlesher@digium.com> - - * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged - revisions 109775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 | - tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines - Change back to using ldap_initialize() and let the user specify a - URL directly, instead of trying to piece it together, badly. - ........ - -2008-03-18 21:03 +0000 [r109716] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar - 2008) | 20 lines Merged revisions 109713 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar - 2008) | 12 lines This patch makes it so that all queue member - status changes are handled through device state code. This - removes several problems people were seeing where their queue - members would get into an "unknown" state. Huge props go to atis - on this one since he was the one who found the code section that - was causing the problem and proposed the solution. I just wrote - what he suggested :) (closes issue #12127) Reported by: atis - Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested - by: atis, jvandal ........ ................ - -2008-03-18 20:14 +0000 [r109684] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) - | 4 lines Set protocol version, port number correctly. (closes - issue #12211, closes issue #12209) Reported by: sylvain ........ - -2008-03-18 19:24 +0000 [r109654] Jason Parker <jparker@digium.com> - - * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | - 15 lines Merged revisions 109648 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | - 7 lines Allow codecs that use log2comp (g726) to compile - correctly on x86 with gcc4 optimizations. (closes issue #12253) - Reported by: fossil Patches: log2comp.patch uploaded by fossil - (license 140) ........ ................ - -2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500 - (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar - 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a - NULL channel closes issue #12242 Reported by Yourname ........ - ................ - - * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar - 2008) | 3 lines Add format attribute to printf-style functions in - astmm.h ........ - -2008-03-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta6 released. - -2008-03-18 17:01 +0000 [r109546] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar - 2008) | 3 lines Add format attribute to printf-style functions in - astmm.h ........ - -2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming <kpfleming@digium.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar - 2008) | 2 lines fix up various warnings found via the addition of - format string checking... some of these were really, really bad - code ........ - -2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant <russell@digium.com> - - * Makefile, channels/chan_misdn.c, include/asterisk/strings.h, - res/res_indications.c, utils/extconf.c, main/asterisk.c, - apps/app_voicemail.c, utils/check_expr.c, - cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /, - res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c, - utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c, - include/asterisk/astobj.h, res/res_agi.c, main/features.c, - apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, - include/asterisk/utils.h, channels/chan_sip.c, - apps/app_festival.c, main/translate.c, main/jitterbuf.c, - utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged - revisions 109447 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 | - twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines - Go through and fix a bunch of places where character strings were - being interpreted as format strings. Most of these changes are - solely to make compiling with -Wsecurity and -Wformat=2 happy, - and were not actual problems, per se. I also added format - attributes to any printf wrapper functions I found that didn't - have them. -Wsecurity and -Wmissing-format-attribute added to - --enable-dev-mode. ........ - - * configs/sip_notify.conf.sample, /: Merged revisions 109111 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar - 2008) | 10 lines Add sample events for aastra phones. - aastra-check-cfg is the same as the other check-cfg entries, and - aastra-xml is to load a pre-configured xml script. (closes issue - #12229) Reported by: gowen72 Patches: aastra.patch uploaded by - gowen72 (license 432) ........ - -2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4: - Merged revisions 109451 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 | - kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 - lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work - properly ........ - -2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant <russell@digium.com> - - * main/dial.c, /: Merged revisions 108962 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008) - | 15 lines Merged revisions 108961 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) - | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes - issue #12228) Reported by: andrew Patches: SRC.patch uploaded by - andrew (license 240) ........ ................ - -2008-03-18 15:16 +0000 [r109398] Joshua Colp <jcolp@digium.com> - - * main/manager.c, /, main/logger.c: Merged revisions 109396 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar - 2008) | 3 lines Make sure values are interpreted as character - strings and not format strings. (AST-2008-004) ........ - -2008-03-18 15:14 +0000 [r109397] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ael-ntest23 (added), - pbx/ael/ael-test/ael-ntest23/t1/a.ael, - pbx/ael/ael-test/ael-ntest23/t1/b.ael, - pbx/ael/ael-test/ael-ntest23/t1/c.ael, - pbx/ael/ael-test/ael-ntest23/t2/d.ael, - pbx/ael/ael-test/ael-ntest23/t2/e.ael, - pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c, - pbx/ael/ael-test/ref.ael-ntest23 (added), - pbx/ael/ael-test/ael-ntest23/t3/g.ael, - pbx/ael/ael-test/ael-ntest23/t3/h.ael, - pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex, - pbx/ael/ael-test/ael-ntest23/t3/j.ael, - pbx/ael/ael-test/ael-ntest23/qq.ael, - pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2, - pbx/ael/ael-test/ael-ntest23/t3, /, - pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions - 109357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | - 25 lines Merged revisions 109309 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | - 17 lines (closes issue #11903) Reported by: atis Many thanks to - atis for spotting this problem and reporting it. The fix was to - straighten out how items are placed on and removed from the file - stack. Regressions as well as the provided test case helped to - straighten out all code paths. valgrind was used to make sure all - memory allocated was freed. Sorry for not solving this earlier. I - got distracted. Added the ntest23 regression test, which is - mainly a copy of ntest22, but with a few juicy errors thrown in, - to replicate the kind of error that atis spotted. ........ - ................ - -2008-03-18 15:11 +0000 [r109395] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 | - qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do - not return with a successful authentication if the From header - ends up empty. (AST-2008-003) ........ - -2008-03-18 15:09 +0000 [r109392] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, - 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 - lines Put a maximum limit on the number of payloads accepted, and - also make sure a given payload does not exceed our maximum value. - (AST-2008-002) ........ ................ - -2008-03-18 00:40 +0000 [r109283] Sean Bright <sean.bright@gmail.com> - - * /, configure, configure.ac: Merged revisions 109282 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar - 2008) | 1 line Fix a typo ........ - -2008-03-17 22:24 +0000 [r109254] Terry Wilson <twilson@digium.com> - - * build_tools/cflags.xml, /, build_tools/menuselect-deps.in, - configure, include/asterisk/autoconfig.h.in, main/Makefile, - configure.ac, main/http.c, main/minimime (removed), - build_tools/make_buildopts_h, makeopts.in: Merged revisions - 109229 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 | - twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines - Replace minimime with superior GMime library so that the entire - contents of an http post are not read into memory. This does - introduce a dependency on the GMime library for handling HTTP - POSTs, but it is available in most distros. If the library is - present, then the compile flag for ENABLE_UPLOADS is enabled by - default in menuselect. ........ - -2008-03-17 22:07 +0000 [r109228] Mark Michelson <mmichelson@digium.com> - - * /, main/utils.c: Merged revisions 109227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar - 2008) | 20 lines Merged revisions 109226 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar - 2008) | 12 lines Fix a logic flaw in the code that stores lock - info which is displayed via the "core show locks" command. The - idea behind this section of code was to remove the previous lock - from the list if it was a trylock that had failed. Unfortunately, - instead of checking the status of the previous lock, we were - referencing the index immediately following the previous lock in - the lock_info->locks array. The result of this problem, under the - right circumstances, was that the lock which we currently in the - process of attempting to acquire could "overwrite" the previous - lock which was acquired. While this does not in any way affect - typical operation, it *could* lead to misleading "core show - locks" output. ........ ................ - -2008-03-17 18:11 +0000 [r109175] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008) - | 11 lines Update the directory of placed calls on skinny phones - when dialing a channel that does not provide progress (analog ZAP - lines) The phone does handle the double update on calls to - channels that do provide progress and wont insert duplicate items - (closes issue #12239) Reported by: DEA Patches: - chan_skinny-call-log.txt uploaded by DEA (license 3) ........ - -2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /, configure, configure.ac, acinclude.m4: Merged - revisions 109166 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 | - kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3 - lines don't define Zaptel features as libraries, they aren't, and - we don't want '--with-zaptel-<foo>' configure options for them - also some minor cleanups ........ - -2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) | - 12 lines Merged revisions 109107 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 - lines 200 OKs in response to a reinvite need to be sent reliably. - If the remote side does not receive one the dialog will be torn - down. (closes issue #12208) Reported by: atrash ........ - ................ - -2008-03-17 14:21 +0000 [r109027] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar - 2008) | 14 lines Merged revisions 109012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar - 2008) | 6 lines Make sure that we release the lock on the spyee - channel if the spyee or spy has hung up (closes issue #12232) - Reported by: atis ........ ................ - -2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008) - | 7 lines Fix polling for mailbox changes in mailboxes that are - not in the default vm context. (closes issue #12223) Reported by: - DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3) - ........ - -2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 108799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 | - russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines - Make sure configure is run before menuselect on a clean checkout - (closes issue #12197) Reported by: juggie Patches: 12197.diff - uploaded by juggie (license 24) ........ - - * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008) - | 13 lines Merged revisions 108796 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) - | 5 lines Fix a channel name issue. chan_oss registers the - "Console" channel type, but it created channels with an "OSS" - prefix. (closes issue #12194, reported by davidw, patched by me) - ........ ................ - - * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r108793 | russell | 2008-03-14 15:04:56 -0500 - (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) - | 4 lines Update the SuSE init script to start networking before - asterisk, as well. (closes issue #12200, reported by and change - suggested by reinerotto) ........ ................ - - * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008) - | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we - have all the required libs reported by the tool. (closes issue - #12067, reported by Juggie, patched by me) ........ - -2008-03-14 16:54 +0000 [r108739] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar - 2008) | 41 lines Merged revisions 108737 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar - 2008) | 33 lines Fix a race condition in the SIP packet scheduler - which could cause a crash. chan_sip uses the scheduler API in - order to schedule retransmission of reliable packets (such as - INVITES). If a retransmission of a packet is occurring, then the - packet is removed from the scheduler and retrans_pkt is called. - Meanwhile, if a response is received from the packet as - previously transmitted, then when we ACK the response, we will - remove the packet from the scheduler and free the packet. The - problem is that both the ACK function and retrans_pkt attempt to - acquire the same lock at the beginning of the function call. This - means that if the ACK function acquires the lock first, then it - will free the packet which retrans_pkt is about to read from and - write to. The result is a crash. The solution: 1. If the ACK - function fails to remove the packet from the scheduler and the - retransmit id of the packet is not -1 (meaning that we have not - reached the maximum number of retransmissions) then release the - lock and yield so that retrans_pkt may acquire the lock and - operate. 2. Make absolutely certain that the ACK function does - not recursively lock the lock in question. If it does, then - releasing the lock will do no good, since retrans_pkt will still - be unable to acquire the lock. (closes issue #12098) Reported by: - wegbert (closes issue #12089) Reported by: PTorres Patches: - 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested - by: jvandal ........ ................ - -2008-03-14 14:33 +0000 [r108684] Jason Parker <jparker@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r108683 | qwell | 2008-03-14 09:32:55 -0500 - (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) | - 4 lines Fix a potential segfault if chan (or chan->music_state) - is NULL. Closes issue #12210, credit to edantie for pointing this - out. ........ ................ - -2008-03-13 21:48 +0000 [r108587] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 108586 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 | - mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3 - lines Make this compile ........ - -2008-03-13 21:41 +0000 [r108585] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, main/channel.c, /, - include/asterisk/channel.h: Merged revisions 108584 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r108584 | russell | 2008-03-13 16:40:43 -0500 - (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) - | 11 lines Fix another issue that was causing crashes in chanspy. - This introduces a new datastore callback, called chan_fixup(). - The concept is exactly like the fixup callback that is used in - the channel technology interface. This callback gets called when - the owning channel changes due to a masquerade. Before this was - introduced, if a masquerade happened on a channel being spyed on, - the channel pointer in the datastore became invalid. (closes - issue #12187) (reported by, and lots of testing from atis) (props - to file for the help with ideas) ........ ................ - -2008-03-13 21:31 +0000 [r108582] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 108529 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 | - mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 - lines Fixing a potential buffer overflow in the manager command - ModuleCheck. Though this overflow is exploitable remotely, we are - NOT issuing a security advisory for this since in order to - exploit the overflow, the attacker would have to establish an - authenticated manager session AND have the system privilege. By - gaining this privilege, the attacker already has more powerful - weapons at his disposal than overflowing a buffer with a - malformed manager header, so the vulnerability in this case - really lies with the authentication method that allowed the - attacker to gain the system privilege in the first place. - ........ - -2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) - | 18 lines Merged revisions 108530 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) - | 10 lines Make a tweak that gets the LEDs on polycom phones to - blink when an extension that has been subscribed to goes on hold. - Otherwise, they just stay on like it does when an extension is in - use. (closes issue #11263) Reported by: russell Patches: - notify_hold.rev1.txt uploaded by russell (license 2) Tested by: - russell ........ ................ - - * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008) - | 2 lines Fix a place where configuration values could cause an - overflow of a buffer. ........ - - * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008) - | 12 lines Merged revisions 108469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) - | 4 lines Fix a couple uses of sprintf. The second one could - actually cause an overflow of a stack buffer. It's not a security - issue though, it only depends on your configuration. ........ - ................ - - * /, main/features.c: Merged revisions 107465 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 | - file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines - Clarify comment about masquerading and playback of the parking - slot. (closes issue #12180) Reported by: davidw ........ - - * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 | - file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If - we receive a 488 on a T38 request reinvite back to audio. As well - reinvite across a bridge back to audio if one side doesn't - negotiate to T38. (closes issue #8677) Reported by: alex-911 - ........ - - * /: Merged revisions 106892 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 | - mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line - Make sure we don't start a call when we have already done so in - response to a COT message ........ - - * /, main/editline/Makefile.in: Merged revisions 106843 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r106843 | qwell | 2008-03-07 16:15:20 -0600 - (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | - 5 lines Fix hardcoded grep in editline, were GNU grep is - required. (closes issue #12124) Reported by: dmartin ........ - ................ - - * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, - channels/chan_sip.c, res/res_phoneprov.c, main/http.c, - include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008) - | 3 lines Rename ast_tcptls_server_instance to session_instance, - since this pertains to server and client usage. ........ - - * /, main/http.c: Merged revisions 108346 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 | - russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines - Make the default prefix empty, like it was in Asterisk 1.4. - (closes issue #12198, reported by bkruse, patched by me) ........ - -2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 | - mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3 - lines Let's get this to compile ........ - - * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar - 2008) | 22 lines Merged revisions 108288 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar - 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for - autocongestion in chan_sip. The scheduler callback will always - return 0. This means that this id is never rescheduled, so it - makes no sense to loop trying to delete the id from the scheduler - queue. If we fail to remove the item from the queue once, it will - fail every single time. (Yes I realize that in this case, the - macro would exit early because the id is set to -1 in the - callback, but it still makes no sense to use that macro in favor - of calling ast_sched_del once and being done with it) This is the - first of potentially several such fixes. ........ - ................ - - * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500 - (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar - 2008) | 12 lines Added a large comment before the AST_SCHED_DEL - macro to explain its purpose as well as when it is appropriate - and when it is not appropriate to use it. I also removed the part - of the debug message that mentions that this is probably a bug - because there are some perfectly legitimate places where - ast_sched_del may fail to delete an entry (e.g. when the - scheduler callback manually reschedules with a new id instead of - returning non-zero to tell the scheduler to reschedule with the - same idea). I also raised the debug level of the debug message in - AST_SCHED_DEL since it seems like it could come up quite - frequently since the macro is probably being used in several - places where it shouldn't be. Also removed the redundant line, - file, and function information since that is provided by ast_log. - ........ ................ - -2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar - 2008) | 14 lines Merged revisions 108086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar - 2008) | 6 lines if we receive an INVITE with a Content-Length - that is not a valid number, or is zero, then don't process the - rest of the message body looking for an SDP closes issue #11475 - Reported by: andrebarbosa ........ ................ - -2008-03-12 19:59 +0000 [r108138] Russell Bryant <russell@digium.com> - - * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r108137 | russell | 2008-03-12 14:59:05 -0500 - (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) - | 40 lines (closes issue #12187, reported by atis, fixed by me - after some brainstorming on the issue with mmichelson) - Update - copyright info on app_chanspy. - Fix a race condition that caused - app_chanspy to crash. The issue was that the chanspy datastore - magic that was used to ensure that spyee channels did not - disappear out from under the code did not completely solve the - problem. It was actually possible for chanspy to acquire a - channel reference out of its datastore to a channel that was in - the middle of being destroyed. That was because datastore - destruction in ast_channel_free() was done near the end. So, this - left the code in app_chanspy accessing a channel that was - partially, or completely invalid because it was in the process of - being free'd by another thread. The following sort of shows the - code path where the race occurred: - ============================================================================= - Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) - --------------------------------------||------------------------------------- - ast_channel_free() || - remove channel from channel list || - - lock/unlock the channel to ensure || that no references retrieved - from || the channel list exist. || - --------------------------------------||------------------------------------- - || channel_spy() - destroy some channel data || - Lock chanspy - datastore || - Retrieve reference to channel || - lock channel || - - Unlock chanspy datastore - --------------------------------------||------------------------------------- - - destroy channel datastores || - call chanspy datastore d'tor || - which NULL's out the ds' || - Operate on the channel ... - reference to the channel || || - free the channel || || || - - unlock the channel - --------------------------------------||------------------------------------- - ============================================================================= - ........ ................ - -2008-03-12 18:31 +0000 [r108085] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h, - main/audiohook.c: Merged revisions 108084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | - 12 lines Merged revisions 108083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 - lines Add a trigger mode that triggers on both read and write. - The actual function that returns the combined audio frame though - will wait until both sides have fed in audio, or until one side - stops (such as the case when you call Wait). (closes issue - #11945) Reported by: xheliox ........ ................ - -2008-03-12 17:03 +0000 [r108033] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 108032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008) - | 12 lines Merged revisions 108031 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) - | 4 lines Destroy the channel lock after the channel datastores. - (inspired by issue #12187) ........ ................ - -2008-03-12 07:44 +0000 [r107879-107999] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 | - tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines - Deadlock fixes (closes issue #12143) Reported by: kactus Patches: - 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: kactus ........ - - * main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged - revisions 107960 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 | - tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines - Revert several changes from revision 102525, as the changes were - not compatible, and, in fact, introduced regressions. (Closes - issue #12190) ........ - - * contrib/scripts/iax-friends.sql, /, - contrib/scripts/sip-friends.sql: Merged revisions 107878 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500 - (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008) - | 2 lines Document all of the possible realtime fields ........ - ................ - -2008-03-11 23:38 +0000 [r107828] Jason Parker <jparker@digium.com> - - * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r107827 | qwell | 2008-03-11 18:38:00 -0500 - (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | - 7 lines Update documentation for pgsql ODBC voicemail. (closes - issue #12186) Reported by: jsmith Patches: - vm_pgsql_doc_update.patch uploaded by jsmith (license 15) - ........ ................ - -2008-03-11 22:59 +0000 [r107723-107793] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_sqlite.c, main/config.c, res/res_config_curl.c, - res/res_config_pgsql.c, res/res_config_odbc.c, /, - include/asterisk/config.h, res/res_config_ldap.c: Merged - revisions 107791 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 | - tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines - An offhand comment from Russell made me realize that the - configuration file caching would not work properly for users.conf - and any other file read from more than one place. I needed to add - the filename which requested the config file to get it to work - properly. ........ - -2008-03-11 20:54 +0000 [r107720] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 107718 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | - 13 lines Merged revisions 107714 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | - 5 lines Copy voicemail dependency logic for res_adsi to - chan_gtalk and chan_jingle (for jabber). (closes issue #12014) - Reported by: junky ........ ................ - -2008-03-11 20:51 +0000 [r107716] Kevin P. Fleming <kpfleming@digium.com> - - * /, Makefile.rules, channels/Makefile: Merged revisions 107715 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500 - (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar - 2008) | 2 lines get chan_vpb to build properly in dev mode - ........ ................ - -2008-03-11 20:37 +0000 [r107584-107711] Joshua Colp <jcolp@digium.com> - - * /, apps/app_page.c: Merged revisions 107710 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 | - file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines - Dial a device even if it's state is unknown. (closes issue - #12184) Reported by: bluecrow76 Patches: - asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by - bluecrow76 (license 270) ........ - - * /, main/features.c: Merged revisions 107659 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) | - 12 lines Merged revisions 107646 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 - lines Make sure the visible indication is on the right channel so - when the masquerade happens the proper indication is enacted. - (closes issue #11707) Reported by: iam ........ ................ - - * /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | - 12 lines Merged revisions 107637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 - lines Add an additional check for setting conference parameter - when using the marked user options. It was possible for it to - return to a no listen/no talk state if a masquerade happened. - (closes issue #12136) Reported by: aragon ........ - ................ - -2008-03-11 15:39 +0000 [r107374-107526] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar - 2008) | 2 lines fix another potential bug found by gcc 4.3 - ........ - - * apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /, - apps/app_sms.c: Merged revisions 107466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar - 2008) | 10 lines Merged revisions 107464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar - 2008) | 2 lines fix various other problems found by gcc 4.3 - ........ ................ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - apps/app_sms.c: Merged revisions 107462 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar - 2008) | 10 lines Merged revisions 107461 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar - 2008) | 2 lines stop checking for mktime() in the configure - script... we don't use it, and the test is buggy under gcc 4.3 - ........ ................ - - * /, configure, main/Makefile, configure.ac, makeopts.in: Merged - revisions 107409 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar - 2008) | 13 lines Merged revisions 107408 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar - 2008) | 5 lines check for compiler support for - -fno-strict-overflow before using it (tested with Debian's gcc - 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview - ........ ................ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 107406 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar - 2008) | 10 lines Merged revisions 107405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar - 2008) | 2 lines fix small bug in IMAP toolkit testing ........ - ................ - - * main/udptl.c, utils/Makefile, /, main/Makefile, - main/editline/readline.c, res/Makefile: Merged revisions 107373 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500 - (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar - 2008) | 11 lines fix up various compiler warnings found with - gcc-4.3: - the output of flex includes a static function called - 'input' that is not used, so for the moment we'll stop having the - compiler tell us about unused variables in the flex source files - (a better fix would be to improve our flex post-processing to - remove the unused function) - main/stdtime/localtime.c makes - assumptions about signed integer overflow, and gcc-4.3's improved - optimizer tries to take advantage of handling potential overflow - conditions at compile time; for now, suppress these optimizations - until we can fiure out if the code needs improvement - - main/udptl.c has some references to uninitialized variables; in - one case there was no bug, but in the other it was certainly - possibly for unexpected behavior to occur - - main/editline/readline.c had an unused variable ........ - ................ - -2008-03-11 01:27 +0000 [r107336] Terry Wilson <twilson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008) - | 10 lines Merged revisions 107290 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) - | 2 lines If we fail to alloc a channel, we should re-lock the - pvt structure before returning. ........ ................ - -2008-03-10 23:46 +0000 [r107289] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 107019 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 | - murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way - back in July, in r.75706, a fix was made ot the strftime usages, - which was good, but in this case, the check for a nil time was - accidentally removed, and now it is restored, to keep timevals - like '1969-12-31 17:00:00' from showing up in the cdrs. No idea - what databases will do with this. No bugs filed as yet, but it - felt like a bug. ........ - -2008-03-10 20:29 +0000 [r107180] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) | - 13 lines Merged revisions 107173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | - 5 lines Make sure to reenable echo can after a "failed" - (canceled, etc) three-way call. (closes issue #11335) Reported - by: rebuild ........ ................ - -2008-03-10 20:18 +0000 [r107101-107163] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 107162 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) - | 16 lines Merged revisions 107161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) - | 8 lines Fix another bug specifically related to asynchronous - call origination. Once the PBX is started on the channel using - ast_pbx_start(), then the ownership of the channel has been - passed on to another thread. We can no longer access it in this - code. If the channel gets hung up very quickly, it is possible - that we could access a channel that has been free'd. (inspired by - BE-386) ........ ................ - - * main/pbx.c, /: Merged revisions 107159 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) - | 17 lines Merged revisions 107158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) - | 9 lines Fix some bugs related to originating calls. If the code - failed to start a PBX on the channel (such as if you set a call - limit based on the system's load average), then there were cases - where a channel that has already been free'd using ast_hangup() - got accessed. This caused weird memory corruption and crashes to - occur. (fixes issue BE-386) (much debugging credit goes to - twilson, final patch written by me) ........ ................ - - * main/channel.c, /: Merged revisions 107103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008) - | 10 lines Merged revisions 107102 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) - | 2 lines Resolve a compiler warning. ........ ................ - - * main/channel.c, /: Merged revisions 107100 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008) - | 11 lines Merged revisions 107099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) - | 3 lines Fix a race condition where the generator can go away - (closes issue #12175, reported by edantie, patched by me) - ........ ................ - -2008-03-10 15:46 +0000 [r107069] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 107068 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 | - mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10 - lines app_queue has now been doxygenified thanks to snuffy! The - ony thing I changed was the way that locks are referenced, since - the old 1.2 names were still used in the comments. (closes issue - #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff - uploaded by snuffy (license 35) ........ - -2008-03-10 14:38 +0000 [r107018] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged - revisions 107017 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) | - 15 lines Merged revisions 107016 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 - lines Move where unanswered CDRs are dropped to the CDR core, not - everything uses app_dial. (closes issue #11516) Reported by: ys - Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested - by: anest, jcapp, dartvader ........ ................ - -2008-03-08 17:54 +0000 [r106997] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we don't start a call on a channel - that has already started a call - -2008-03-08 16:14 +0000 [r106947] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar - 2008) | 10 lines Merged revisions 106945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar - 2008) | 2 lines don't generate D-Channel "up" and "down" messages - unless the channel state is actually changing; also, generate the - "up" message when an implicit "up" occurs due to reception of a - normal event when we thought the channel was "down" ........ - ................ - -2008-03-07 22:53 +0000 [r106897] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008) - | 10 lines Merged revisions 106895 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) - | 2 lines Only start the SLA thread if SLA has actually been - configured. ........ ................ - -2008-03-07 19:34 +0000 [r106790] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 106789 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) | - 12 lines Merged revisions 106788 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 - lines Ignore source update control frame. (closes issue #12168) - Reported by: plack ........ ................ - -2008-03-07 17:18 +0000 [r106686-106713] Russell Bryant <russell@digium.com> - - * /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r106707 | russell | 2008-03-07 11:17:30 -0600 - (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) - | 8 lines Change a warning message to a debug message. This is - happening quite frequently, and it is not worth spamming users - with these messages unless we are pretty confident that it should - never happen. As it stands today, it _will_ and _does_ happen and - until that gets cleaned up a reasonable amount on the development - side, let's not spam the logs of everyone else. (closes issue - #12154) ........ ................ - - * doc/smdi.txt, /: Merged revisions 106684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 | - russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines - fix example usage ........ - -2008-03-07 16:27 +0000 [r106554-106662] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600 - (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008) - | 3 lines Warn the user when a temporary greeting exists (Closes - issue #11409) ........ ................ - - * main/rtp.c, /: Merged revisions 106607 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008) - | 11 lines Merged revisions 106606 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) - | 3 lines Properly initialize rtp->schedid (Closes issue #12154) - ........ ................ - - * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, - apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, - funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /, - channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c, - utils/extconf.c: Merged revisions 106553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008) - | 14 lines Merged revisions 106552 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) - | 6 lines Safely use the strncat() function. (closes issue - #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt - uploaded by Corydon76 (license 14) ........ ................ - -2008-03-07 01:19 +0000 [r106502-106520] Russell Bryant <russell@digium.com> - - * doc/smdi.txt, /: Merged revisions 106518 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 | - russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line - minor text changes ........ - - * doc/smdi.txt, /: Merged revisions 106507 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 | - russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines - Add updated SMDI documentation that I had only sitting in my - email ... oops ........ - - * main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c, - main/file.c: Merged revisions 106501 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 | - russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines - Merge changes from team/russell/g722-sillyness ... Fix a number - of other places where the number of samples in a G722 frame was - not properly handled because of various reasons. main/rtp.c: - - When a G722 frame is read from the smoother, the number of - samples in the frame must be divided by 2 before being sent out - over the network. Even though G722 is 16 kHz, an error in some - previous spec has made it so that we have to list the number of - samples such as if it was 8 kHz. main/file.c: - When scheduling - the next time to expect a frame, take into account that the - format of the file we're reading from may not be 8 kHz. - codecs/codec_g722.c: - When converting from G722 to slinear, - g722_decode() expects its samples parameter to be in the silly - (real samples / 2) format. Make it so. - When converting from - slinear to G722, properly set the number of samples in the frame - to be the number of bytes of output * 2. formats/format_pcm.c: - - This format module handles G722, among a number of other formats. - However, the read() and seek() functions did not account for the - fact that G722 has 2 samples per byte. (closes issue #12130, - reported by rickross, patched by me) ........ - -2008-03-06 22:16 +0000 [r106442] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, /: Merged revisions 106438 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar - 2008) | 16 lines Merged revisions 106437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar - 2008) | 8 lines Quell an annoying message that is likely to print - every single time that ast_pbx_outgoing_app is called. The reason - is that __ast_request_and_dial allocates the cdr for the channel, - so it should be expected that the channel will have a cdr on it. - Thanks to joetester on IRC for pointing this out ........ - ................ - -2008-03-06 22:15 +0000 [r106440] Jason Parker <jparker@digium.com> - - * /, main/file.c: Merged revisions 106439 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 | - qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines - Fix file playback in many cases. (closes issue #12115) Reported - by: pj Patches: v2-fileexists.patch uploaded by dimas (license - 88) (with modifications by me) Tested by: dimas, qwell, russell - ........ - -2008-03-06 20:39 +0000 [r106433] Donny Kavanagh <donnyk@gmail.com> - - * /, res/res_agi.c: Merged revisions 106399 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 | - juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines - trivial fix for an agi error when attempting to use EAGI on a - dead/hungup channel, we now print an error that makes sense given - our removal of deadagi as an actual application. (closes issue - #12161) Reported by: explidous Patches: res_agi_12161.patch - uploaded by juggie (license 24) Tested by: juggie ........ - -2008-03-06 05:25 +0000 [r106330-106359] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008) - | 7 lines Missing braces, fix parsing (closes issue #12112) - Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008 - uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76 - ........ - - * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600 - (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) - | 2 lines Upgrade to the next release of sounds ........ - ................ - -2008-03-06 00:23 +0000 [r106299-106320] Russell Bryant <russell@digium.com> - - * channels/chan_oss.c, main/rtp.c, main/channel.c, - channels/chan_phone.c, main/dial.c, channels/chan_skinny.c, - main/file.c, channels/chan_h323.c, channels/chan_alsa.c, - include/asterisk/frame.h, channels/chan_mgcp.c, - channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /, - channels/chan_sip.c, channels/chan_console.c, - apps/app_followme.c: Merged revisions 106239 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | - 12 lines Merged revisions 106235 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 - lines Add a control frame to indicate the source of media has - changed. Depending on the underlying technology it may need to - change some things. (closes issue #12148) Reported by: jcomellas - ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r106238 | russell | 2008-03-05 16:40:58 -0600 - (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) - | 3 lines Fix a potential deadlock and a few different potential - crashes. (closes issue #12145, reported by thiagarcia, patched by - me) ........ ................ - - * /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008) - | 7 lines document var_metric usage to prevent bugreports that - are actually configuration issues (closes issue #12151) Reported - by: caio1982 Patches: DB_metric3.diff uploaded by caio1982 - (license 22) ........ - - * main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged - revisions 105933 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) - | 13 lines Merged revisions 105932 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) - | 5 lines Fix a bug that I just noticed in the RTP code. The - calculation for setting the len field in an ast_frame of audio - was wrong when G.722 is in use. The len field represents the - number of ms of audio that the frame contains. It would have set - the value to be twice what it should be. ........ - ................ - - * funcs/func_global.c, /: Merged revisions 105899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 | - russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines - Fix the SHARED() read callback to properly unlock the channel. - This function could not have worked, as it left the channel - locked in all cases. ........ - - * main/manager.c, /: Merged revisions 105864 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 | - mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5 - lines There are several places in manager.c where BUFSIZ is used - for a buffer which will contain nowhere near that amount of data. - This makes these buffers more reasonably sized. ........ - - * main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c, - apps/app_queue.c: Merged revisions 105841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 | - tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines - Fix minor misuses of snprintf ........ - - * main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c, - main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c, - main/frame.c, /, main/say.c, main/utils.c, main/enum.c, - main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c, - main/loader.c, main/term.c, main/cli.c, main/channel.c, - main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c, - main/features.c, main/logger.c, main/app.c, main/image.c, - main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c: - Merged revisions 105840 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 | - tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines - Whitespace changes only ........ - - * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, - main/http.c, include/asterisk/tcptls.h: Merged revisions 105804 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04 - Mar 2008) | 2 lines add a destroy API call for a server instance - ........ - - * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, - main/http.c, include/asterisk/tcptls.h: Merged revisions 105785 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04 - Mar 2008) | 2 lines More public API name changes to use an - appropriate ast_ prefix ........ - - * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, - channels/chan_sip.c, res/res_phoneprov.c, main/http.c, - include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008) - | 2 lines Rename public object server_instance to - ast_tcptls_server_instance ........ - - * /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 | - russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines - Fix some bugs in the SIP tcp helper thread. - fix a spot where a - lock wouldn't get unlocked in an error condition - call - ast_mutex_destroy() on the lock before freeing its memory - (related to issue #11972) ........ - - * /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 | - twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines - Set username to default to the category name if it isn't - overridden by a usernmae= setting in users.conf ........ - - * main/rtp.c, /: Merged revisions 105677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) | - 10 lines Merged revisions 105676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 - lines In addition to setting the marker bit let's change our ssrc - so they know for sure it is a different source. ........ - ................ - - * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: - Merged revisions 105675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | - 16 lines Merged revisions 105674 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 - lines When a new source of audio comes in (such as music on hold) - make sure the marker bit gets set. (closes issue #10355) Reported - by: wdecarne Patches: 10355.diff uploaded by file (license 11) - (closes issue #11491) Reported by: kanderson ........ - ................ - -2008-03-05 17:42 +0000 [r106140] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008) - | 3 lines Should check these values for non-NULL before scanning. - (Closes issue #12147) ........ - -2008-03-05 15:43 +0000 [r106041] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar - 2008) | 15 lines Merged revisions 106038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar - 2008) | 7 lines when a PRI call must be moved to a different B - channel at the request of the other endpoint, ensure that any DSP - active on the original channel is moved to the new one (closes - issue #11917) Reported by: mavetju Tested by: mavetju ........ - ................ - -2008-03-05 15:31 +0000 [r106037] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, include/asterisk/sched.h: Merged - revisions 106036 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008) - | 15 lines Merged revisions 106015 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) - | 7 lines Correctly initialize retransid in SIP, and ensure that - the warning when failing to delete a schedule entry can actually - hit the log. (closes issue #12140) Reported by: slavon Patches: - sch2.patch uploaded by slavon (license 288) (Patch slightly - modified by me) ........ ................ - -2008-03-04 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta5 released. - -2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant <russell@digium.com> - - * CHANGES: Update CHANGES heading - - * funcs/func_version.c: Simplify a trivial snprintf() with - ast_copy_string() - - * main/hashtab.c: Make it so you don't have to cast away const in a - couple places - - * main/hashtab.c: remove unnecessary casts - - * main/pbx.c: - Add curly braces around the while loop - Properly - break out of the loop on error when an included context is not - found - - * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use - sizeof() instead of a magic number - - * channels/chan_zap.c: Fix some code that was improperly changed in - revision 104866 from issue #12079. (closes issue #12129, reported - by elguero, patched by me) - -2008-03-03 18:08 +0000 [r105573] Jason Parker <jparker@digium.com> - - * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) | - 7 lines Fix types for astNumChannels and astConfigCallsProcessed. - (closes issue #12114) Reported by: jeffg Patches: 12114.patch - uploaded by jeffg (license 192) ........ - -2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, /: Merged revisions 105570 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03 - Mar 2008) | 3 lines In the case of an ast_channel allocation - failure, take the local_pvt out of the pvt list before destroying - it. ........ - - * channels/chan_local.c, /: Merged revisions 105568 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 - Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt - struct when ast_channel allocation fails. Also, in passing, - centralize the code necessary to destroy a local_pvt. ........ - - * main/autoservice.c, /: Merged revisions 105565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008) - | 3 lines Update the copyright information for autoservice. Most - of the code in this file now is stuff that I have written - recently ... ........ - - * main/channel.c, main/autoservice.c, /, - include/asterisk/_private.h, main/asterisk.c: 3) In addition to - merging the changes below, change trunk back to a regular LIST - instead of an RWLIST. The way this list works makes it such that - a RWLIST provides no additional benefit. Also, a mutex is needed - for use with the thread condition. Merged revisions 105563 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) - | 24 lines Merge in some changes from - team/russell/autoservice-nochans-1.4 These changes fix up some - dubious code that I came across while auditing what happens in - the autoservice thread when there are no channels currently in - autoservice. 1) Change it so that autoservice thread doesn't keep - looping around calling ast_waitfor_n() on 0 channels twice a - second. Instead, use a thread condition so that the thread - properly goes to sleep and does not wake up until a channel is - put into autoservice. This actually fixes an interesting bug, as - well. If the autoservice thread is already running (almost always - is the case), then when the thread goes from having 0 channels to - have 1 channel to autoservice, that channel would have to wait - for up to 1/2 of a second to have the first frame read from it. - 2) Fix up the code in ast_waitfor_nandfds() for when it gets - called with no channels and no fds to poll() on, such as was the - case with the previous code for the autoservice thread. In this - case, the code would call alloca(0), and pass the result as the - first argument to poll(). In this case, the 2nd argument to - poll() specified that there were no fds, so this invalid pointer - shouldn't actually get dereferenced, but, this code makes it - explicit and ensures the pointers are NULL unless we have valid - data to put there. (related to issue #12116) ........ - -2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 105560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 - lines It is possible for no audio to pass between the current - digit and next digit so expand logic that clears emulation to - AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg - Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: - tbsky ........ - - * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6 - lines Add a comment to describe some logic. (closes issue #12120) - Reported by: flefoll Patches: - chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license - 244) ........ - -2008-03-01 03:59 +0000 [r105509] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Add support for 16KHz signed linear. - -2008-03-01 02:03 +0000 [r105479] Tilghman Lesher <tlesher@digium.com> - - * /: Drop bad property - -2008-03-01 01:30 +0000 [r105477] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, include/asterisk/app.h, - main/global_datastores.c, /, main/features.c, main/app.c, - include/asterisk/global_datastores.h: Asterisk, when parking can - drop rights a caller when a parking timeout occurs. Also, when - doing built-in attended transfers, sometimes incorrectly passes - rights from the transferrer to the transferee. This patch tries - to fixes the parking issue and lays some groundwork for later - fixing the transfer issue. (closes issue #11520) Reported by: - pliew Tested by: otherwiseguy - -2008-03-01 00:53 +0000 [r105461] Russell Bryant <russell@digium.com> - - * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI - command to control custom device states. Also, do some additional - code cleanup and improvement in passing. (closes issue #12106) - Reported by: nizon Patches: devstate-patch.txt uploaded by nizon - (license 415) -- Updated to trunk, and tab completion added by me - -2008-02-29 23:53 +0000 [r105411] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Convert to use ast_str - -2008-02-29 23:36 +0000 [r105410] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: Merged revisions 105409 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) - | 23 lines Fix a major bug in autoservice. There was a race - condition in the handling of the list of channels in autoservice. - The problem was that it was possible for a channel to get removed - from autoservice and destroyed, while the autoservice thread was - still messing with the channel. This led to memory corruption, - and caused crashes. This explains multiple backtraces I have seen - that have references to autoservice, but do to the nature of the - issue (memory corruption), could cause crashes in a number of - areas. (fixes the crash in BE-386) (closes issue #11694) (closes - issue #11940) The following issues could be related. If you are - the reporter of one of these, please update to include this fix - and try again. (potentially fixes issue #11189) (potentially - fixes issue #12107) (potentially fixes issue #11573) (potentially - fixes issue #12008) (potentially fixes issue #11189) (potentially - fixes issue #11993) (potentially fixes issue #11791) ........ - -2008-02-29 18:34 +0000 [r105378] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample: Add documentation for setting - username/password in SIP dial string. (closes issue #11587) - Reported by: sobomax Patches: dialstring_doc.diff uploaded by - sobomax (license 359) - -2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008) - | 1 line Fix a potential memory leak ........ - - * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c: - Remove unnecessary if statements before calling iks_delete - (redundant check is done inside iks_delete), thus making the code - conform with coding guidelines. - -2008-02-29 13:55 +0000 [r105262] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb - 2008) | 4 lines Bump up the size of the uniqueid variable. - (closes issue #12107) Reported by: asgaroth ........ - -2008-02-29 13:12 +0000 [r105210] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Automatically create new buddy upon reception - of a presence stanza of type subscribed. (closes issue #12066) - Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by - phsultan (license 73) trunk-12066-1.diff uploaded by phsultan - (license 73) Tested by: ffadaie, phsultan - -2008-02-29 01:15 +0000 [r105176] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008) - | 7 lines Update init script for LSB compat (closes issue #9843) - Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by - ibc (license 211) Tested by: paravoid ........ - -2008-02-28 22:39 +0000 [r105144] Russell Bryant <russell@digium.com> - - * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c: - Merged revisions 105116 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) - | 8 lines Fix a bug in the lock tracking code that was discovered - by mmichelson. The issue is that if the lock history array was - full, then the functions to mark a lock as acquired or not would - adjust the stats for whatever lock is at the end of the array, - which may not be itself. So, do a sanity check to make sure that - we're updating lock info for the proper lock. (This explains the - bizarre stats on lock #63 in BE-396, thanks Mark!) ........ - -2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb - 2008) | 6 lines When using autofill, members who are in use - should be counted towards the number of available members to call - if ringinuse is set to yes. Thanks to jmls who brought this issue - up on IRC ........ - - * main/dial.c, /: Merged revisions 104841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb - 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels - a lockable list. This is because in some cases, the ast_dial may - exist in multiple threads due to asynchronous execution of its - application, and I found some cases where race conditions could - exist. 2. Check in ast_dial_join to be sure that the channel - still exists before attempting to lock it, since it could have - gotten hung up but the is_running_app flag on the - ast_dial_channel may not have been cleared yet. (closes issue - #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by - putnopvut (license 60) Tested by: jvandal ........ - -2008-02-28 19:21 +0000 [r105006] Jason Parker <jparker@digium.com> - - * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb - 2008) | 9 lines Make pbx_exec pass an empty string into - applications, if we get NULL. This protects against possible - segfaults in applications that may try to use data before - checking length (ast_strdupa'ing it, for example) (closes issue - #12100) Reported by: foxfire Patches: 12100-nullappargs.diff - uploaded by qwell (license 4) ........ - -2008-02-28 14:42 +0000 [r104974] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_vpb.cc: Fix crash when configuration does not match - hardware detection. (closes issue #12096) Reported by: mmickan - Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) - -2008-02-28 04:37 +0000 [r104921] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb - 2008) | 2 lines According to a video at www.cisco.com, the 7921G - supports 6 line appearances. ........ - -2008-02-28 00:11 +0000 [r104869] Tilghman Lesher <tlesher@digium.com> - - * /, main/Makefile, build_tools/strip_nonapi: Merged revisions - 104868 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) - | 7 lines Compatibility fix for PPC64 (closes issue #12081) - Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch - uploaded by jcollie (license 412) Tested by: jcollie, Corydon76 - ........ - -2008-02-27 23:58 +0000 [r104866] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: reduce indentation in alloc_sub (issue - #12079) Reported by: tzafrir Patches: alloc_sub uploaded by - tzafrir (license 46) - -2008-02-27 21:02 +0000 [r104788] Joshua Colp <jcolp@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2 - lines Don't loop around infinitely trying to spy on our own - channel, and don't forget to free/detach the datastore upon - hangup of the spy. ........ - -2008-02-27 20:37 +0000 [r104784] Mark Michelson <mmichelson@digium.com> - - * /, main/file.c: Merged revisions 104783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb - 2008) | 4 lines Bump a couple of more buffers up by 2 so that - annoying warnings aren't generated like crazy on every - fileexists_core call. ........ - -2008-02-27 19:36 +0000 [r104756] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in - favor of 'val', which serves the exact same purpose. - -2008-02-27 18:20 +0000 [r104705] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /: Merged revisions 104704 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008) - | 2 lines Ensure the session ID can't be 0. ........ - -2008-02-27 17:45 +0000 [r104687] Joshua Colp <jcolp@digium.com> - - * /, main/file.c: Merged revisions 104665 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 - lines Bump up the buffer by 2. ........ - -2008-02-27 17:36 +0000 [r104643] Russell Bryant <russell@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008) - | 4 lines Fix a problem in ChanSpy where it could get stuck in an - infinite loop without being able to detect that the calling - channel hung up. (closes issue #12076, reported by junky, patched - by me) ........ - -2008-02-27 17:31 +0000 [r104617] Jason Parker <jparker@digium.com> - - * /, main/features.c: Merged revisions 104598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) | - 8 lines Inherit language from the transfering channel on a blind - transfer. (closes issue #11682) Reported by: caio1982 Patches: - local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22) - Tested by: caio1982, victoryure ........ - -2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp <jcolp@digium.com> - - * /, main/loader.c: Merged revisions 104596 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4 - lines Use the lock (which already existed, it just wasn't used) - on the updaters list to protect the contents instead of the - overall module list lock. (closes issue #12080) Reported by: - ChaseVenters ........ - - * channels/chan_sip.c: After further discussion revert my previous - commit for this. Currently in order to ensure devicestate is the - expected value in another module (such as app_queue) then - chan_sip must be loaded before hand. - -2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/file.c: Merged revisions 104593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb - 2008) | 8 lines fallback to standard English prompts properly - when using new prompt directory layout (closes issue #11831) - Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG - (license 20) (modified by me to improve code and conform rest of - function to coding guidelines) ........ - -2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: When queueing up a device state change when - the peer is loaded from the configuration give it a state of not - in use. We have to do this because the channel technology may not - yet be registered so the state could not be queried and would be - considered invalid. (closes issue #12087) Reported by: liorm - - * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4 - lines Only stop the MWI monitor thread if it was actually - started. (closes issue #12086) Reported by: francesco_r ........ - -2008-02-27 15:34 +0000 [r104534] Tilghman Lesher <tlesher@digium.com> - - * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is - specified. (Closes issue #12083) - -2008-02-27 15:31 +0000 [r104533] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression - introduced by state changes. (closes issue #12078) Reported by: - dimas Patches: v1-12078.patch uploaded by dimas (license 88) - (closes issue #12074) Reported by: Ivan - -2008-02-27 08:20 +0000 [r104502] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_vpb.cc, configs/vpb.conf.sample, - include/asterisk/module.h: Bring Voicetronix driver up to date - with current drivers (closes issue #12084) Reported by: mmickan - Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) - module.h.diff uploaded by mmickan (license 400) vpb.conf.sample - uploaded by mmickan (license 400) - -2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant <russell@digium.com> - - * doc/janitor-projects.txt: note that the chan_sip conversion is - already in progress - - * doc/janitor-projects.txt: add another janitor project - - * doc/janitor-projects.txt: Add the stuff from the janitor projects - page that is still relevant. I figure that if we keep this in the - tree, it will be much easier to keep up to date. The page on - asterisk.org just links to this on svn.digium.com/view - -2008-02-27 03:52 +0000 [r104418] Jason Parker <jparker@digium.com> - - * doc/janitor-projects.txt (added): Create placeholder file...for - now. - -2008-02-27 02:05 +0000 [r104388] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Whitespace changes only - -2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant <russell@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008) - | 3 lines Avoid some recursion in the cleanup code for the - chanspy datastore (closes issue #12076, reported by junky, - patched by me) ........ - -2008-02-26 22:14 +0000 [r104301] Steve Murphy <murf@digium.com> - - * res/snmp/agent.c: small change to allow this file to compile. No - problem if you don't install the libsnmp package. - -2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant <russell@digium.com> - - * main/asterisk.c: I swear I compiled this ... *cough* - - * res/res_phoneprov.c: fix this module, too - - * funcs/func_version.c: fix this module - - * Makefile, include/asterisk, build_tools/make_version_h (added): - Re-add the automatically generated version.h, so that modules can - include for making build time decisions for cross asterisk - version compatibility - - * main/manager.c, channels/chan_sip.c, include/asterisk/version.h - (removed), build_tools/make_version_c, res/res_agi.c, - main/http.c, include/asterisk/ast_version.h (added): Rename - version.h to ast_version.h. Next, I will be re-adding version.h - as an automatically generated file like it used to be. This still - needs to be there for modules that have to check it to compile - against multiple asterisk versions. - -2008-02-26 19:14 +0000 [r104215] Joshua Colp <jcolp@digium.com> - - * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an - 'e' option to ResetCDR which re-enables a CDR that has been - disabled. (closes issue #11170) Reported by: kratzers Patches: - ResetCDR.1.diff uploaded by kratzers (license 307) - -2008-02-26 18:40 +0000 [r104176] Tilghman Lesher <tlesher@digium.com> - - * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while, - even around single statements. 2) Revise the argument parsing - section, showing use of the standard macros. 3) Fix a typo. - -2008-02-26 18:27 +0000 [r104140-104142] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 104141 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) | - 1 line Add badshell to .PHONY target (thanks Kevin) ........ - - * Makefile, /: Merged revisions 104139 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) | - 2 lines Since all shells aren't as awesome as bash, we have to - fail if somebody tries to use a literal "~" in DESTDIR. ........ - -2008-02-26 16:51 +0000 [r104137] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Formatting and doxygen while waiting on an - airport... - -2008-02-26 16:36 +0000 [r104133-104136] Jason Parker <jparker@digium.com> - - * /, sounds/Makefile: Merged revisions 104135 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) | - 5 lines Revert previous abspath change. ...abspath is new in GNU - make 3.81. I feel so...defeated. Must find new fix! ........ - - * /, sounds/Makefile: Merged revisions 104132 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) | - 9 lines Fix a very bizarre issue we were seeing with our buildbot - when using a DESTDIR that wasn't an absolute path (such as - DESTDIR=~/asterisk-1.4). Apparently what was happening, was that - some of the targets were being expanded to the full path, so $@ - ended up being /root/asterisk-1.4/[...]/ rather than - ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" - in GNU make. (*cough* - http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) - ........ - -2008-02-26 14:51 +0000 [r104127] Mark Michelson <mmichelson@digium.com> - - * main/features.c: Remove more hardcoded pipe symbols and replace - with commas. (closes issue #12072) Reported by: SimonSharman - Patches: features.patch uploaded by SimonSharman (license 410) - Tested by: SimonSharman - -2008-02-26 06:43 +0000 [r104125] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c: Use the readhandle for reads (closes issue - #12069) - -2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant <russell@digium.com> - - * res/res_smdi.c: Add a \todo to convert this module to the event - system - - * CHANGES: Update CHANGES for SMDI stuff - - * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample, - include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions - 104119 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) - | 33 lines Merge changes from team/russell/smdi-1.4 This commit - brings in a significant set of changes to the SMDI support in - Asterisk. There were a number of bugs in the current - implementation, most notably being that it was very likely on - busy systems to pop off the wrong message from the SMDI message - queue. So, this set of changes fixes the issues discovered as - well as introducing some new ways to use the SMDI support which - are required to avoid the bugs with grabbing the wrong message - off of the queue. This code introduces a new interface to SMDI, - with two dialplan functions. First, you get an SMDI message in - the dialplan using SMDI_MSG_RETRIEVE() and then you access - details in the message using the SMDI_MSG() function. A side - benefit of this is that it now supports more than just chan_zap. - For example, with this implementation, you can have some FXO - lines being terminated on a SIP gateway, but the SMDI link in - Asterisk. Another issue with the current implementation is that - it is quite common that the station ID that comes in on the SMDI - link is not necessarily the same as the Asterisk voicemail box. - There are now additional directives in the smdi.conf - configuration file which let you map SMDI station IDs to Asterisk - voicemail boxes. Yet another issue with the current SMDI support - was related to MWI reporting over the SMDI link. The current code - could only report a MWI change when the change was made by - someone calling into voicemail. If the change was made by some - other entity (such as with IMAP storage, or with a web interface - of some kind), then the MWI change would never be sent. The SMDI - module can now poll for MWI changes if configured to do so. This - work was inspired by and primarily done for the University of - Pennsylvania. (also related to issue #9260) ........ - -2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option - in favor of dialplan substring variable syntax. (closes issue - #12060) - - * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008) - | 10 lines This patch fixes some pretty significant problems with - how app_chanspy handles pointers to channels that are being spied - upon. It was very likely that a crash would occur if the channel - being spied upon hung up. This was because the current - ast_channel handling _requires_ that the object is locked or else - it could disappear at any time (except in the owning channel - thread). So, this patch uses some channel datastore magic on the - spied upon channel to be able to detect if and when the channel - goes away. (closes issue #11877) (patch written by me, but thanks - to kpfleming for the idea, and to file for review) ........ - - * /, main/utils.c: Merged revisions 104102 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008) - | 7 lines Improve the lock tracking code a bit so that a bunch of - old locks that threads failed to lock don't sit around in the - history. When a lock is first locked, this checks to see if the - last lock in the list was one that was failed to be locked. If it - is, then that was a lock that we're no longer sitting in a - trylock loop trying to lock, so just remove it. (inspired by - issue #11712) ........ - -2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be - saved in Postgres. Note that these changes are - backward-compatible, so no changes to UPGRADE.txt are necessary. - (closes issue #9279) Reported by: rottenroddy Patches: - 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76 - - * funcs/func_global.c: Shared space for variables (instead of - letting other channels muck with your own) (closes issue #11943) - Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt - uploaded by Corydon76 (license 14) Tested by: jmls - - * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25 - Feb 2008) | 5 lines If the destination folder is full, don't - delete a message when exiting. (closes issue #12065) Reported by: - selsky Patch by: (myself) ........ - -2008-02-25 21:40 +0000 [r104096] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 - lines Make it so a users.conf user creates both a SIP peer and a - SIP user. The user will be used for inbound authentication for - the device, and peer will be used for placing calls to the - device. (closes issue #9044) Reported by: queuetue Patches: - sip-gui-friend.diff uploaded by qwell (license 4) ........ - -2008-02-25 20:50 +0000 [r104093] Jason Parker <jparker@digium.com> - - * /, main/config.c: Merged revisions 104092 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) | - 11 lines Allow the use of #include and #exec in situations where - the max include depth was only 1. Specifically, this fixes using - #include and #exec in extconfig.conf. This was basically caused - because the config file itself raises the include level to 1. I - opted not to raise the include limit, because recursion here - could cause very bizarre behavior. Pointed out, and tested by - jmls (closes issue #12064) ........ - -2008-02-25 19:02 +0000 [r104089] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Instead of outputting a verbose message - every so often let's make it a debug message. - -2008-02-25 19:00 +0000 [r104088] Brett Bryant <bbryant@digium.com> - - * doc/siptls.txt, configs/sip.conf.sample: Adding more tls - configuration details to sip.conf sample, with a list of valid - ciphers provided in both files. .. First commit since July, woot - -2008-02-25 18:38 +0000 [r104087] Russell Bryant <russell@digium.com> - - * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25 - Feb 2008) | 4 lines Ensure that the channel doesn't disappear in - agent_logoff(). If it does, it could cause a crash. (fixes the - crash reported in BE-396) ........ - -2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 - lines If a resubscription comes in for a dialog we no longer know - about tell the remote side that the dialog does not exist so they - subscribe again using a new dialog. (closes issue #10727) - Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file - (license 11) ........ - - * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 - lines Due to recent changes tag will no longer be NULL if not - present so we have to use ast_strlen_zero to see if it's actually - blank. (closes issue #12061) Reported by: flefoll Patches: - chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll - (license 244) ........ - - * res/res_config_pgsql.c: Fix building of trunk. dbpass is always - going to exist. - -2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: Enforce a space between function args as per - code review. - - * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned - on, and pgsql installed, I get warnings that stops the compile. - They are fixed now. - -2008-02-22 23:56 +0000 [r104045] Doug Bailey <dbailey@digium.com> - - * channels/chan_zap.c, configure, configure.ac: Add protection to - chan_zap build when NEONMWI events are not defined - -2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher <tlesher@digium.com> - - * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES, - include/asterisk/manager.h: Move Originate to a separate - privilege and require the additional System privilege to call out - to a subshell. - - * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008) - | 6 lines Backwards debug message. (closes issue #12052) Reported - by: flefoll Patches: chan_sip.c.br14.patch_found-notfound - uploaded by flefoll (license 244) ........ - - * res/res_config_pgsql.c: Allow database password to be NULL and - several other cleanups. (closes issue #12048) Reported by: bukaj - Patches: 20080222__bug12048.diff.txt uploaded by Corydon76 - (license 14) Tested by: bukaj - -2008-02-21 21:27 +0000 [r104031] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix a typo - -2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson <mmichelson@digium.com> - - * res/res_agi.c: Instead of a notice, make the message about a - hung-up channel a debug message, and revert the original logic on - the if statement. Thanks to Juggie for bringing this to my - attention. - -2008-02-21 17:38 +0000 [r104024] Doug Bailey <dbailey@digium.com> - - * channels/chan_zap.c: Added configuration distinction between neon - and fsk mwi detection Add the detection for neon MWI events got - rid of extraneous handle_init_event call in monitor loop - -2008-02-21 16:46 +0000 [r104020] Mark Michelson <mmichelson@digium.com> - - * res/res_agi.c: Don't print the fact that we are using dead mode - in AGI if called from the 'h' extension since it is well-known - that it will be running in dead mode. (closes issue #12046) - Reported by: explidous - -2008-02-21 16:44 +0000 [r104019] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - Disable epoll as it has caused more obscure issues then any of my - previous code. I will continue to work on it in a separate branch - to make it stable for a release and test it against the following - issues. (closes issue #11253) Reported by: falves11 (closes issue - #11657) Reported by: davevg (closes issue #11033) Reported by: - falves11 - -2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c, /: Merged revisions 104015 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb - 2008) | 2 lines reduce the likelihood that HTTP Manager session - ids will consist of primarily '1' bits ........ - -2008-02-21 05:21 +0000 [r104014] Tilghman Lesher <tlesher@digium.com> - - * utils/astman.c: Ignore some more unused generated events. (closes - issue #12042) Reported by: junky Patches: astman_events.diff - uploaded by junky (license 177) - -2008-02-20 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta4 released. - -2008-02-20 22:34 +0000 [r103957] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb - 2008) | 8 lines Clear up confusion when viewing the - QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the - user's perspective, the queue does exist, we shouldn't tell them - we couldn't find the queue. Instead since it is a dead queue, - report a 0 waiting count This issue was brought up on IRC by jmls - ........ - -2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Try to do Packet2Packet bridging with - chan_h323 if reinviting isn't enabled. (closes issue #11901) - Reported by: pj - - * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6 - lines Don't wait for additional digits when overlap dialing is - enabled if the setup message contains the sending_complete - information element. (closes issue #11785) Reported by: klaus3000 - Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt - uploaded by klaus3000 (license 65) ........ - -2008-02-20 21:41 +0000 [r103908] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: Merged revisions 103904 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed, - 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL - while attempting to lock it. (closes issue #12039) Reported by: - danpwi ........ - -2008-02-20 21:36 +0000 [r103903] Jason Parker <jparker@digium.com> - - * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone - detection routines. Separate fax detection from digit detected. - Added CED (called) tone detection for fax (previously, only CNG - (calling) was supported). Separate DTMF/MF code paths where - appropriate. Allow detection of arbitary tones. (closes issue - #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch - uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache - -2008-02-20 21:08 +0000 [r103902] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix a crash due to the wrong variable being - used when building a directory string. (closes issue #12027) - Reported by: jaroth Patches: forward.patch uploaded by jaroth - (license 50) Tested by: jaroth - -2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/sched.h: Add some documentation fixups - - * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20 - Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022) - Reported by: asgaroth Patches: 20080219__bug12022.diff.txt - uploaded by Corydon76 (license 14) Tested by: asgaroth ........ - -2008-02-20 15:21 +0000 [r103844] Mark Michelson <mmichelson@digium.com> - - * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't - been converted to ',' (closes issue #12034) Reported by: kowalma - -2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp <jcolp@digium.com> - - * main/audiohook.c: *mumble* - - * main/audiohook.c: file not found. - - * main/audiohook.c: Minor test... - -2008-02-20 00:49 +0000 [r103833] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: When using IMAP storage, if the folder you - attempt to save to does not exist, create it first. (closes issue - #12032) Reported by: jaroth Patches: createfolder.patch uploaded - by jaroth (license 50) Tested by: jaroth - -2008-02-19 22:35 +0000 [r103831-103832] Jason Parker <jparker@digium.com> - - * main/channel.c: Make sure to mask out non-audio first as well - - * main/channel.c: Maybe we should set the value before we test it? - Fixes an issue people have been seeing (unreported?) with file - playback not working. - -2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp <jcolp@digium.com> - - * main/loader.c: Add a log message that appears when you try to - unload a module that isn't loaded. (closes issue #12033) Reported - by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by - jamesgolovich (license 176) - - * main/file.c: Only output a log message saying the format does not - exist if it actually does not exist, not if the file itself could - not be opened. (closes issue #11828) Reported by: IgorG Patches: - readfile.v1.diff uploaded by IgorG (license 20) - - * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6 - lines Send CallerID Name in setup message. (closes issue #11241) - Reported by: tusar Patches: h323id_as_callerid_name.patch - uploaded by tusar (license 344) ........ - -2008-02-19 20:06 +0000 [r103822] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, /: Merged revisions 103821 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 - Feb 2008) | 8 lines Account for the fact that the "other" channel - can disappear while the local pvt is not locked. (fixes a problem - introduced in rev 100581) (closes issue #12012) Reported by: - stevedavies Patch by me ........ - -2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp <jcolp@digium.com> - - * apps/app_authenticate.c: len already contains the position we - want to examine, if we move one left again we'll actually - probably be looking at a digit. (issue #12030) Reported by: - alligosh - - * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add - CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan - application. This will either be set to NOCHANNEL if the given - channel was not found or SUCCESS if it worked. (closes issue - #11553) Reported by: johan Patches: - UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) - CHANGES.channelredirect.patch uploaded by johan (license 334) - app_channelredirect-20080219.patch uploaded by johan (license - 334) - -2008-02-19 18:14 +0000 [r103818] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm - Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm - (license 99) Patch fixes problem of device state incorrectly - reporting idle before PBX answers incoming call on FXO channel. - Device status is updated now during new channel creation. - -2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp <jcolp@digium.com> - - * /, configure, configure.ac: Merged revisions 103812 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb - 2008) | 4 lines Don't look for launchd when cross compiling. - (closes issue #12029) Reported by: ovi ........ - -2008-02-19 00:59 +0000 [r103805] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: Change verbosity into debug for Hebrew (and various - whitespace fixes) (Closes issue #12011) - -2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 103801 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) | - 10 lines Ensure that emulated DTMFs do not get interrupted by - another begin frame. (closes issue #11740) Reported by: gserra - Patches: v1-11740.patch uploaded by dimas (license 88) (closes - issue #11955) Reported by: tsearle (closes issue #10530) Reported - by: xmarksthespot ........ - - * main/channel.c, main/frame.c, channels/chan_sip.c, - include/asterisk/channel.h, include/asterisk/frame.h: Add a - non-invasive API for application level manipulation of T38 on a - channel. This uses control frames (so they can even pass across - IAX2) to negotiate T38 and provided a way of getting the current - status of T38 using queryoption. This should by no means cause - any issues and if it does I will take responsibility for it. - (closes issue #11873) Reported by: dimas Patches: - v4-t38-api.patch uploaded by dimas (license 88) - - * main/frame.c: Add some missing control frames. - -2008-02-18 22:33 +0000 [r103796] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) | - 1 line Fix previous commit so that we actually disable - echocanbridged if echocancel is off. ........ - -2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Commit chan_zap portion of #11964: add the - ability to get ORIG_CALLED_NUM - -2008-02-18 21:30 +0000 [r103791] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) | - 4 lines Correct a message when echocancelwhenbridged is on, but - echocancel is not. Closes issue #12019 ........ - -2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes - in. Also add support for multiple DPCs per linkset. #11779 - -2008-02-18 20:53 +0000 [r103787] Mark Michelson <mmichelson@digium.com> - - * /, main/app.c: Merged revisions 103786 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb - 2008) | 10 lines There was an invalid assumption when calculating - the duration of a file that the filestream in question was - created properly. Unfortunately this led to a segfault in the - situation where an unknown format was specified in voicemail.conf - and a voicemail was recorded. Now, we first check to be sure that - the stream was written correctly or else assume a zero duration. - (closes issue #12021) Reported by: jakep Tested by: putnopvut - ........ - -2008-02-18 19:47 +0000 [r103783] Michiel van Baak <michiel@vanbaak.info> - - * main/asterisk.c: make the output of 'core show settings' a bit - nicer. (closes issue #12020) Reported by: seanbright Patches: - asterisk.c.patch uploaded by seanbright (license 71) - -2008-02-18 17:45 +0000 [r103781] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) - | 9 lines When a SIP channel is being auto-destroyed, it's - possible for it to still be in bridge code. When that happens, we - crash. Delay the RTP destruction until the bridge is ended. - (closes issue #11960) Reported by: norman Patches: - 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: norman ........ - -2008-02-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta3 released. - -2008-02-18 17:12 +0000 [r103772] Olle Johansson <oej@edvina.net> - - * main/channel.c, channels/chan_sip.c: Make sure we can set up - calls without audio (text+video). And ... it works! - -2008-02-18 16:40 +0000 [r103771] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb - 2008) | 10 lines Fix a linked list corruption that under the - right circumstances could lead to a looped list, meaning it will - traverse forever. (closes issue #11818) Reported by: michael-fig - Patches: 11818.patch uploaded by putnopvut (license 60) Tested - by: michael-fig ........ - -2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp <jcolp@digium.com> - - * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h: - Add an API call (ast_async_parseable_goto) which parses a goto - string and does an async goto instead of an explicit goto. - (closes issue #11753) Reported by: johan - - * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2 - lines Don't care if the extension given doesn't exist for - subscription based MWI. ........ - -2008-02-18 10:10 +0000 [r103755] Olle Johansson <oej@edvina.net> - - * CHANGES, channels/chan_iax2.c: - No space in manager event names, - please - Add new event to CHANGES - -2008-02-18 04:43 +0000 [r103754] Tilghman Lesher <tlesher@digium.com> - - * build_tools/cflags.xml, main/channel.c, main/pbx.c, - funcs/func_channel.c, include/asterisk/channel.h, CHANGES, - main/cli.c: Context tracing for channels (closes issue #11268) - Reported by: moy Patches: - chantrace-datastored-encapsulated-rev94934.patch uploaded by moy - (license 222) - -2008-02-16 21:22 +0000 [r103750] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now - monitoring chan_skinny port with nagios or zabbix wont generate - noise on the console. @ok tilghman - -2008-02-15 23:32 +0000 [r103742] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15 - Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race - condition (closes issue #11780) Reported by: guillecabeza - Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license - 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license - 380) ........ - -2008-02-15 23:20 +0000 [r103740] Mark Michelson <mmichelson@digium.com> - - * CHANGES: Document GotoIfTime change from svn revision 103738 - -2008-02-15 23:14 +0000 [r103739] Russell Bryant <russell@digium.com> - - * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related - to the use of AES encryption. 1024 was used instead of 128 when - using AES from OpenSSL. Many thanks to d1mas for figuring this - one out! (closes issue #11946) Reported by: bbhoss Patches: - v1-11946.patch uploaded by dimas (license 88) - -2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: Add proper "false" case behavior to GotoIfTime - (closes issue #11719) Reported by: kshumard Patches: - gotoiftime.twobranches.patch uploaded by kshumard (license 92) - Tested by: kshumard - - * apps/app_voicemail.c: Fix redeclaration of variables when using - IMAP storage (closes issue #11988) Reported by: jaroth Patches: - variable_cleanup.patch uploaded by jaroth (license 50) - -2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant <russell@digium.com> - - * /, main/loader.c: Merged revisions 103728 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) - | 4 lines In the case that you try to directly reload a module - has returned AST_MODULE_LOAD_DECLINE, log a message indicating - that the module is not fully initialized and must be initialized - using "module load". ........ - - * /, main/loader.c: Merged revisions 103726 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) - | 6 lines Don't attempt to execute the reload callback for a - module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash - that was reported against chan_console in trunk. (closes issue - #11953, reported by junky, fixed by me) ........ - -2008-02-15 17:32 +0000 [r103725] Mark Michelson <mmichelson@digium.com> - - * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged - revisions 103722 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb - 2008) | 8 lines Final round of changes for configure script logic - for IMAP Now if a directory is specified, then we will search - that directory for a source installation of the IMAP toolkit. If - none is found, then we will use that directory as the basis for - detecting a package installation of the IMAP c-client. If that - check fails, then configure will fail. ........ - -2008-02-15 17:29 +0000 [r103723] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c, - include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c, - apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c, - main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h, - main/ast_expr2f.c, channels/misdn/ie.c, - channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c, - include/asterisk/strings.h, res/res_config_ldap.c, - include/asterisk/time.h: Fix up some doxygen issues. (closes - issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy - (license 35) - -2008-02-15 15:45 +0000 [r103716] Tilghman Lesher <tlesher@digium.com> - - * utils/conf2ael.c: Remove extraneous copy (closes issue #12002) - Reported by: junky Patches: conf2ael.diff uploaded by junky - (license 177) - -2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson <mmichelson@digium.com> - - * configure, configure.ac: Merging of changes from 1.4 revision - 103713. - - * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as - made to 1.4 in revision 103710 - - * doc/tex/imapstorage.tex: Trunk version of 1.4's imap - documentation updates - - * configure, configure.ac: See commit message for svn revision - 103698. This behavior is the same as what is described there. The - difference is that trunk already had the --with-imap=system - option, but it only checked the include path for headers in the - imap directory and not also the c-client directory. - -2008-02-14 21:21 +0000 [r103694] Jason Parker <jparker@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify - ldap autoconf function, so that an incorrect ldap library is not - found on Solaris (it is incompatible). Also removes second check - for awk, which causes Solaris to find an incompatible version of - awk. (closes issue #11829) Reported by: snuffy Patches: - bug-11829.diff uploaded by snuffy (license 35) - -2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu, - 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........ - - * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu, - 14 Feb 2008) | 9 lines Fix the new message count if delete=yes - when using IMAP storage. (closes issue #11406) Reported by: - jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license - 50) Tested by: jaroth ........ - - * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change - the queue holdtime announcement to happen at any interval (not - just greater than two minutes). Remove the saying of less-than - for holdtime announcements since it can lead to awkward holdtime - announcements. Using '1' as a queue-round-seconds value is no - longer valid. (closes issue #9736) Reported by: caio1982 Patches: - queue_announce5.diff uploaded by caio1982 (license 22) Tested by: - caio1982, putnopvut - -2008-02-14 19:52 +0000 [r103685] Jason Parker <jparker@digium.com> - - * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) | - 5 lines Document the 'l' option to the CDR() function. (Thanks - voipgate for pointing out the option, and Leif for providing text - for it.) Closes issue #11695. ........ - -2008-02-14 19:47 +0000 [r103682] Jeff Peeler <jpeeler@digium.com> - - * apps/app_externalivr.c: a few syntax changes and safer code - -2008-02-14 18:39 +0000 [r103677] Jason Parker <jparker@digium.com> - - * channels/chan_iax2.c: Add periodic jitter stats to CLI and - manager. (closes issue #8188) Reported by: stevedavies Patches: - jblogging-trunk.patch uploaded by stevedavies - jblogging-trunk_wmgrevent.patch uploaded by johann8384 - updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384 - (license 190) (with additional changes by me) Tested by: - stevedavies, johann8384 - -2008-02-14 10:19 +0000 [r103668] Olle Johansson <oej@edvina.net> - - * res/res_agi.c, apps/app_externalivr.c: Formatting fixes - -2008-02-13 21:04 +0000 [r103662] Jeff Peeler <jpeeler@digium.com> - - * apps/app_externalivr.c: (closes issue #11825) Reported by: - ctooley Patches: additional_eivr_commands.patch uploaded by - ctooley (license 136) Tested by: ctooley - -2008-02-13 15:47 +0000 [r103658] Mark Michelson <mmichelson@digium.com> - - * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold - and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold - (closes issue #11904) Reported by: dimas Patches: v2-moh.patch - uploaded by dimas (license 88) Tested by: dimas - -2008-02-13 00:55 +0000 [r103559] Mark Michelson <mmichelson@digium.com> - - * main/event.c: Fix a small logic error in ast_event_iterator_next. - The previous logic allowed for the iterator to indicate there was - more data than there really was, causing the iterator read beyond - the end of the event structure. This led to invalid memory reads - and potential crashes. - -2008-02-12 22:26 +0000 [r103447-103506] Jason Parker <jparker@digium.com> - - * main/manager.c: Even more sane permissions. This should be - handled via a umask, like in many other places. - - * main/manager.c: Use slight more sane permissions - -2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Remove development version notice. - - * main/manager.c: Fix build on *BSD. These permissions constants - are not available there. - -2008-02-12 15:13 +0000 [r103386] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 - lines Even if no CallerID name or number has been provided by the - remote party still use the configured sip.conf ones. (closes - issue #11977) Reported by: pj ........ - -2008-02-12 14:08 +0000 [r103341] Philippe Sultan <philippe.sultan@gmail.com> - - * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags - structure in aji_client and aji_buddy rather than an integer. - Modify calls to various ast_*_flag macros accordingly. - -2008-02-12 00:24 +0000 [r103331] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c, include/asterisk/config.h, CHANGES, - main/config.c: Requested changes from Pari, reviewed by Russell. - Added ability to retrieve list of categories in a config file. - Added ability to retrieve the content of a particular category. - Added ability to empty a context. Created new action to create a - new file. Updated delete action to allow deletion by line number - with respect to category. Added new action insert to add new - variable to category at specified line. Updated action newcat to - allow new category to be inserted in file above another existing - category. - -2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4 - lines If entering a conference with the 'w' option ensure that we - can't listen or speak until the marked user appears. (closes - issue #11835) Reported by: alanmcmillan ........ - - * res/res_agi.c: Remove ast_module_user usage from res_agi. This is - taken care of in the core. - - * main/pbx.c, main/manager.c, main/translate.c, main/logger.c, - main/app.c, main/utils.c, main/indications.c, main/asterisk.c, - main/rtp.c: Just some minor coding style cleanup... - - * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue - #10661) Reported by: junky - -2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming <kpfleming@digium.com> - - * /, configs/zapata.conf.sample: Merged revisions 103315 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb - 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared) - ........ - -2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp <jcolp@digium.com> - - * main/channel.c, channels/chan_iax2.c: Add support for allowing a - native bridge to happen when the L option is enabled. The RTP - bridging could already handle this, it just needed to be enabled - in the main bridging code. (issue #10647) Reported by: samdell3 - - * channels/chan_skinny.c: Change chan_skinny to use debug messages - as appropriate. (closes issue #11967) Reported by: mvanbaak - Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak - (license 7) - -2008-02-11 06:05 +0000 [r103306] James Golovich <james@gnuinter.net> - - * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip - on reload (issue #11930) - -2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix improper indentation. Thanks again to - snuffy for pointing it out. - - * apps/app_queue.c: Add a couple of comments to clarify the - unreffing of queues. Thanks to snuffy for the idea. - - * main/event.c: Fix a problem regarding network vs. host byte order - in the event API. ast_event_iterator_get_ie_type should return - the ie type in host byte order. Furthermore, ast_event_get_ie_raw - should already have its ie type argument in host byte order since - it could be called externally (and it in fact is called in this - way by ast_event_get_cached). - -2008-02-09 11:27 +0000 [r103249] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c, - apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c, - apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c, - apps/app_channelredirect.c, apps/app_directed_pickup.c, - apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c, - apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c: - whitespace fixes only. - -2008-02-09 06:33 +0000 [r103198] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09 - Feb 2008) | 4 lines Commit fix for being unable to send voicemail - from VoiceMailMain Reported by: William F Acker (via the -users - mailing list) Patch by: Corydon76 (license 14) ........ - -2008-02-08 21:26 +0000 [r103171] Russell Bryant <russell@digium.com> - - * main/udptl.c, main/pbx.c, channels/chan_sip.c, - channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c, - main/rtp.c, channels/chan_usbradio.c, main/cdr.c, - channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c, - pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c, - apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c, - apps/app_voicemail.c: Merge changes from - team/mvanbaak/cli-command-audit (closes issue #8925) About a year - ago, as Leif Madsen and Jim van Meggelen were going over the CLI - commands in Asterisk 1.4 for the next version of their book, they - documented a lot of inconsistencies. This set of changes - addresses all of these issues and has been reviewed by Leif. - While this does introduce even more changes to the CLI command - structure, it makes everything consistent, which is the most - important thing. Thanks to all that helped with this one! - -2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes - different arguments in trunk than 1.4. - - * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb - 2008) | 10 lines Prevent a potential three-thread deadlock. Also - added a comment block to explicitly state the locking order - necessary inside app_queue. (closes issue #11862) Reported by: - flujan Patches: 11862.patch uploaded by putnopvut (license 60) - Tested by: flujan ........ - - * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri, - 08 Feb 2008) | 6 lines Yield the thread and return -1 if the - ioctl fails for Zaptel timing device. (closes issue #11891) - Reported by: tzafrir ........ - -2008-02-08 16:49 +0000 [r103044] Russell Bryant <russell@digium.com> - - * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At - the request of ManxPower, include the UPGRADE.txt from 1.2 and - 1.4, as well. This way, if people need to go back and review what - was deprecated in previous major releases, it is readily - available to them. Thanks for the suggestion! - -2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix a network byte order issue and ensure - when creating an outgoing dialog that the socket always contains - information such as type and port. (closes issue #11916) Reported - by: mnnojd - - * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb - 2008) | 4 lines Make sure the presence of dbsecret is factored - into user scoring. (closes issue #11952) Reported by: bbhoss - ........ - -2008-02-07 21:37 +0000 [r102933] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: This is a combination new feature/bug fix for - app_chanspy. New feature: Add the 'e' option, which takes as an - argument a list of interfaces separated by colons. This way, you - will only be able to spy on this limited list of interfaces. Bug - fix: change some pointer checks to ast_strlen_zero so that spying - would work properly even if no channel was specified as the first - argument to chanspy. (closes issue #10072) Reported by: - xmarksthespot Patches: - bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by - xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak - -2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_adsiprog.c: whitespace fixes only - - * apps/app_alarmreceiver.c: There she goes! First commit from me to - trunk \o/ Make app_alarmreceiver honor code guidelines and fix - whitespace errors. No functional changes. - -2008-02-07 20:02 +0000 [r102859] Jason Parker <jparker@digium.com> - - * /, main/features.c: Merged revisions 102858 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) | - 7 lines Specify which digit string was matched in debug message. - (closes issue #11949) Reported by: dimas Patches: - v1-feature-debug.patch uploaded by dimas (license 88) ........ - -2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming <kpfleming@digium.com> - - * /, configs/zapata.conf.sample: Merged revisions 102807 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb - 2008) | 2 lines document usage of 'transfer' configuration option - for ISDN PRI switch-side transfers ........ - -2008-02-06 20:12 +0000 [r102777] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add the channel's unique id to the AgentCalled - manager event to make it more consistent with other manager - events. - -2008-02-06 18:01 +0000 [r102726] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2 - lines Only consider a T.38-only INVITE compatible if we have both - a joint capability between us and them and if they provided T.38. - ........ - -2008-02-06 16:23 +0000 [r102700] Terry Wilson <twilson@digium.com> - - * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY - dialplan functions provided by sergee. I just added the ability - to set multiple fields at once after discussions with Tilghman - and Russell. Currently limited to 30 fields. (closes issue - #11887) Reported by: sergee Patches: - rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy - (license 396) Tested by: sergee, otherwiseguy - -2008-02-06 15:20 +0000 [r102652] Russell Bryant <russell@digium.com> - - * /, configs/features.conf.sample: Merged revisions 102651 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) - | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets - inherited by outbound channels. (due to a discussion between me - and a user via email) ........ - -2008-02-06 03:05 +0000 [r102602] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05 - Feb 2008) | 4 lines Move around some defines to unbreak ODBC - storage. (closes issue #11932) Reported by: snuffy ........ - -2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Remove an extra debug message I left in - - * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, - apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c, - apps/app_followme.c, main/loader.c, channels/chan_usbradio.c, - main/tcptls.c, res/res_agi.c, apps/app_minivm.c, - apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c, - main/astmm.c: Get rid of any remaining ast_verbose calls in the - code in favor of ast_verb (closes issue #11934) Reported by: - mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by - mvanbaak (license 7) - - * apps/app_voicemail.c: Change verbose messages to use the ast_verb - macro. (closes issue #11931) Reported by: snuffy Patches: - bug-11931.diff uploaded by snuffy (license 35) - -2008-02-05 20:51 +0000 [r102500] Jason Parker <jparker@digium.com> - - * main/pbx.c: Change where priority of a goto is adjusted. - Partially reverts 102272. Closes issue #11929 (credit to file for - fix suggestion - we still <3 you) - -2008-02-05 20:03 +0000 [r102454] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue, - 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes - issue #11919) Reported by: eferro Patches: - mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) - Tested by: eferro ........ - -2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot - me. - - * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3 - lines If a REGISTER attempt comes in that is a retransmission of - a previous REGISTER do not create a new nonce value. (issue - #BE-381) ........ - - * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb - 2008) | 4 lines Perform dialing asynchronously when using the - originate CLI command so the CLI does not appear to block. - (closes issue #11927) Reported by: bbhoss ........ - -2008-02-04 21:15 +0000 [r102329] Tilghman Lesher <tlesher@digium.com> - - * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in, - configure.ac, main/asterisk.c: Merged revisions 102323 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008) - | 7 lines Cross-platform fix: OS X now deprecates the use of the - daemon(3) API. (closes issue #11908) Reported by: oej Patches: - 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76 ........ - -2008-02-04 18:39 +0000 [r102297] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Add line numbers to warning/error messages - (and pretty up some existing ones). (closes issue #11894) - Reported by: jmls Patches: chan_zap.patch uploaded by jmls - (license 141) - -2008-02-04 15:16 +0000 [r102272] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Update handling of asyncgoto so it properly works on - channels that are currently executing a PBX. (closes issue - #11914) Reported by: arnd (closes issue #11753) Reported by: - johan - -2008-02-04 14:37 +0000 [r102262] Jason Parker <jparker@digium.com> - - * configs/extensions.ael.sample, configs/extensions.lua.sample: - Change examples to use G here also. Closes issue #11875 - -2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03 - Feb 2008) | 6 lines Missing braces. (closes issue #11912) - Reported by: dimas Patches: sprintf.patch uploaded by dimas - (license 88) ........ - - * main/manager.c: CoreSettings and CoreStatus are missing the - terminating "\r\n". Also, some miscellaneous spacing and - initialization issues. (closes issue #11909) Reported by: srt - Patches: patch-11909-2.diff uploaded by srt (license 378) Tested - by: srt - -2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8 - lines Use the same CSEQ on CANCEL as on INVITE (according to RFC - 3261) (closes issue #9492) Reported by: kryptolus Patches: - bug9492.txt uploaded by oej (license 306) Tested by: oej ........ - - * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 - lines Handle ACK and CANCEL in an invite transaction - even if we - get INFO transactions during the actual call setup. (closes issue - #10567) Reported by: jacksch Tested by: oej Patch by: oej - inspired by suggestions from neutrino88 in the bug tracker - ........ - -2008-02-03 06:43 +0000 [r102064] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Change the version number in the - configure script from 1.4 to 1.6 - -2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant <russell@digium.com> - - * include/asterisk/event.h: The documentation page has to be in its - own comment block to work, apparently. Fix it up! - - * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) - | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that - even though G.722 is 16 kHz, it is supposed to specified as 8 kHz - in the RTP, and RTP timestamps are supposed to be calculated - based on 8 kHz. (Apparently this is due to a bug in a spec, but - people follow it anyway, because it's the spec ...) ........ - -2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01 - Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded - voicemail, and fixed several other bugs while I'm in the area. - (closes issue #11615) Reported by: jamessan Patches: - 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76, jamessan ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 101894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008) - | 2 lines Change detection of getifaddrs to use - AST_C_COMPILE_CHECK, backported from trunk (as suggested by - kpfleming) ........ - - * res/res_config_curl.c: Fix multi, when using the LIKE query. - (closes issue #11889) Reported by: jmls Patches: - res_config_curl.patch uploaded by jmls (license 141) Tested by: - jmls - -2008-02-01 18:24 +0000 [r101869] Jason Parker <jparker@digium.com> - - * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak. - -2008-02-01 18:08 +0000 [r101824] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling - functionality by changing the config file keyword - -2008-02-01 17:44 +0000 [r101823] Jason Parker <jparker@digium.com> - - * /, apps/app_authenticate.c: Move an feof() call to before the - fgets(). This would have exited the loop early if you had an - authentication file with no newline at the end. - -2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant <russell@digium.com> - - * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01 - Feb 2008) | 4 lines Don't overwrite the last character of a line - if it's not a newline. This would happen if the last line in the - file doesn't have a newline. (pointed out by Qwell) ........ - -2008-02-01 16:01 +0000 [r101773] Tilghman Lesher <tlesher@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - main/acl.c: Merged revisions 101772 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008) - | 2 lines Compatibility fix for OpenWRT (reported by Brian - Capouch via the mailing list) ........ - -2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant <russell@digium.com> - - * apps/app_authenticate.c: simplify some code, tweak formatting, - and reduce indentation - - * apps/app_authenticate.c: reduce a level of indentation - - * apps/app_channelredirect.c: Get rid of a goto where there was no - extra cleanup happening at the exit point - - * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 - Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial - strings for the case that no peer or hostname was provided, which - is the one part of the dial string that is absolutely required. - If it's not there, bail out. (closes issue #11897) Reported by - sokhapkin Patch by me ........ - -2008-02-01 00:08 +0000 [r101650] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan - 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle - periods of no channel activity" (closes issue #9256) Reported by: - cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt - uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, - rjain ........ - -2008-01-31 23:14 +0000 [r101611] Russell Bryant <russell@digium.com> - - * /, main/translate.c, main/file.c: Merged revisions 101601 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) - | 12 lines Fix a couple of places where ast_frfree() was not - called on a frame that came from a translator. This showed itself - by g729 decoders not getting released. Since the flag inside the - translator frame never got unset by freeing the frame to indicate - it was no longer in use, the translators never got destroyed, and - thus the g729 licenses were not released. (closes issue #11892) - Reported by: xrg Patches: 11892.diff uploaded by russell (license - 2) Tested by: xrg, russell ........ - -2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Forgot an ! - - * apps/app_queue.c: A change I made to accommodate the "linear" - strategy in trunk caused queue strategies to not be loaded from - realtime queues. This commit fixes that. Thanks to jmls for - pointing this problem out to me on IRC. This also contains some - changes to S_OR where it should be used. Thanks to Qwell for - pointing these out. - -2008-01-31 21:33 +0000 [r101577] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix a simple deadlock that was introduced - _right_ before this code got merged into trunk. (closes issue - #11895, reported by pj, patched by me) - -2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Handle the case of a NULL state_interface when - checking a realtime member. Thanks to jmls for finding this - issue. - - * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan - 2008) | 10 lines 1. Prevent the addition of an extra '/' to the - beginning of an absolute pathname. 2. If ast_monitor_change_fname - is called and the new filename is the same as the old, then exit - early and don't set the filename_changed field in the monitor - structure. Setting it in this case was causing ast_monitor_stop - to erroneously delete them. (closes issue #11741) Reported by: - garlew Tested by: putnopvut ........ - -2008-01-31 19:54 +0000 [r101483] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions - 101482 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) | - 4 lines Solaris compat fixes for struct in_addr funkiness. Issue - #11885, patch by snuffy. ........ - -2008-01-31 19:43 +0000 [r101481] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 101480 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1 - line closes issue #11845; that's the one where there's a 1004 - byte cdr leak with every AMI Redirect to a zap channel ........ - -2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant <russell@digium.com> - - * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31 - Jan 2008) | 2 lines Add more missing locking of the agents list - ... ........ - - * /, channels/chan_agent.c: Merged revisions 101413-101414 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) - | 2 lines Add missing locking to the find_agent() function. - ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 - Jan 2008) | 3 lines Move the locking from find_agent() into the - agent dialplan function handler to ensure that the agent doesn't - disappear while we're looking at it. ........ - -2008-01-31 15:36 +0000 [r101393] Joshua Colp <jcolp@digium.com> - - * funcs/func_realtime.c: Add missing braces. (closes issue #11886) - Reported by: sergee Patches: func_realtime_fix-r101392.diff - uploaded by sergee (license 138) - -2008-01-31 05:28 +0000 [r101373] Russell Bryant <russell@digium.com> - - * CHANGES: remove entry that is no longer in the tree - -2008-01-30 23:10 +0000 [r101344] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: The deprecation of "username" in favor of - "defaultuser" for SIP peers unfortunately broke realtime - configurations which still used the "username" field. This was - taken care of properly when reading from realtime but was not - handled properly when updating a realtime peer. This change also - adds a deprecation NOTICE CLI message that will print if using - the deprecated "username" field. (closes issue #11880) Reported - by: cabal95 Patches: 11880.patch uploaded by putnopvut (license - 60) Tested by: cabal95 - -2008-01-30 20:08 +0000 [r101322] Olle Johansson <oej@edvina.net> - - * configs/cli.conf.sample: Clarify configuration file that can be - misunderstood - -2008-01-30 19:03 +0000 [r101296] Jason Parker <jparker@digium.com> - - * apps/app_controlplayback.c: Allow disabling the default - ffwd/rewind keys in the ControlPlayback application. This is done - in a backward compat way. If the "default" key for ffwd/rew is - used for another option (such as stop), the "default" is removed. - (closes issue #11754) Reported by: johan Patches: - app_controlplayback.c.option3.patch uploaded by johan (license - 334) Tested by: johan, qwell - -2008-01-30 17:12 +0000 [r101271] Olle Johansson <oej@edvina.net> - - * configs/rtppage.conf.sample (removed), apps/app_rtppage.c - (removed): Removing applications that wasn't ready for svn trunk, - as trunk now has pre-release status. - -2008-01-30 16:54 +0000 [r101269] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Make the VoicemailUsersList AMI command - consistent with other manager list functions. (closes issue - #11874) Reported by: srt Patches: voicemail_ami-11847.patch - uploaded by srt (license 378) - -2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson <oej@edvina.net> - - * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change - function to void because it always returned the same value and no - one read it. - - * main/rtp.c: Formatting fixes - -2008-01-30 15:42 +0000 [r101224] Mark Michelson <mmichelson@digium.com> - - * apps/app_rtppage.c: Get trunk to compile - -2008-01-30 15:42 +0000 [r101223] Joshua Colp <jcolp@digium.com> - - * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4 - lines Fix an issue where if a frame of higher sample size - preceeded a frame of lower sample size and ast_slinfactory_read - was called with a sample size of the combined values or higher a - crash would happen. (closes issue #11878) Reported by: stuarth - ........ - -2008-01-30 15:36 +0000 [r101221] Olle Johansson <oej@edvina.net> - - * CHANGES: Update CHANGES with rtppage - -2008-01-30 15:35 +0000 [r101220] Jason Parker <jparker@digium.com> - - * /, configs/extensions.conf.sample: Merged revisions 101219 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37 - -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use - descending channel order of groups, rather than ascending. Fixes - a potential source of confusion in glare-type situations. Issue - 11875, reported by JimVanM. ........ - -2008-01-30 15:30 +0000 [r101218] Olle Johansson <oej@edvina.net> - - * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added): - Add rtppage() application to do multicast or unicast RTP paging - to SIP phones. (closes issue #11797) Reported by: macbrody - Patches: app_rtppage-20080130.c uploaded by macbrody (license - 352) - -2008-01-30 15:27 +0000 [r101217] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan - 2008) | 5 lines Fix a logic error with regards to autofill. Prior - to this change, it was possible for a caller to go out of turn if - autofill were enabled and callers ahead in the queue were - attempting to call a member. This change fixes this. ........ - -2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: simplify this code and eliminate the return - value cast that is no longer necessary - -2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, include/asterisk/channel.h: Constifying the - interface to get pvt_ids in the bridge, based on suggestion from - (const char *) Kevin. Thanks! - - * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7 - lines Stop musiconhold on attended transfer. (closes issue - #11872) Reported by: gareth Patches: svn-101018.patch uploaded by - gareth (license 208) ........ - -2008-01-30 00:58 +0000 [r101126] Jason Parker <jparker@digium.com> - - * CHANGES: Fix a typo - -2008-01-30 00:04 +0000 [r101082] Russell Bryant <russell@digium.com> - - * CHANGES, apps/app_speech_utils.c: Add the 'n' option to - SpeechBackground, which has the application not answer the - channel if it has not already been answered. (closes SPD-51) - -2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, build_tools/make_version: Merged revisions 101080 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29 - Jan 2008) | 1 line updated build_tools to handle the autotag - directory structure changes; changes related to BE-353. Patch by - The Russell and reviewed by The Me. ........ - -2008-01-29 23:02 +0000 [r101036] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan - 2008) | 3 lines Remove a memory leak from updating realtime - queues ........ - -2008-01-29 22:04 +0000 [r101018] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_curl.c: Oops, a sizeof error - -2008-01-29 19:41 +0000 [r100974] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan - 2008) | 6 lines Fixing an erroneous return value returned when - attempting to pause or unpause a queue member fails. Fixes - BE-366, thanks to John Bigelow for writing the patch. ........ - -2008-01-29 17:44 +0000 [r100933] Russell Bryant <russell@digium.com> - - * /, main/Makefile: Merged revisions 100932 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) - | 4 lines Fix the last couple of issues related to building from - a path that contains spaces. (closes issue #11834) ........ - -2008-01-29 17:42 +0000 [r100931] Jason Parker <jparker@digium.com> - - * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan - 2008) | 6 lines Initialize an array to 0s if config option not - specified. (closes issue #11860) Patches: - misdn_get_config.v1.diff uploaded by IgorG (license 20) ........ - -2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 100922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008) - | 3 lines Use GNU make magic instead of shell magic to escape - spaces in the working directory. (related to issue #11834) - ........ - - * Makefile, /: Merged revisions 100882 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008) - | 6 lines Fix building Asterisk when the working path has spaces - in it. (closes issue #11834) Reported by: spendergrass Patched - by: me ........ - -2008-01-29 16:14 +0000 [r100843] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) | - 5 lines Allow zap groups above 30 to work properly. (closes issue - #11590) Reported by: tbsky ........ - -2008-01-29 15:30 +0000 [r100833] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make externip work as documented. If no port - is specified it will use the value of bindport instead of always - being 5060. (closes issue #11858) Reported by: hmodes - -2008-01-29 10:50 +0000 [r100794-100795] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29 - Jan 2008) | 1 line fixed potential segfault in misdn show - channels CLI command ........ - - * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03 - Jan 2008) | 1 line make sure frame is completely clean, before we - send it to asterisk as DTMF. If we don't make it clean, it - happens that one way audio occurs.. ........ - -2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8 - lines (closes issue #11736) Reported by: MVF Patches: - bug11736-2.diff uploaded by oej (license 306) Tested by: oej, - MVF, revolution (russellb: This was the showstopper for the - release.) ........ - - * channels/chan_sip.c: Removing code that wasn't supposed to be - there at all, only at an experimental stage before I found - another solution. Thanks Kevin, for reminding me. - -2008-01-28 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta2 released. - -2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker@digium.com> - - * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added), - doc/tex/channelvariables.tex, makeopts.in: Reintroduce more - chan_vpb stuff that was removed in r100421 and r100422 - -2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson@digium.com> - - * channels/chan_vpb.cc (added), configure, - include/asterisk/autoconfig.h.in, configure.ac, - channels/Makefile: Re-inserting chan_vpb into trunk. - -2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 100675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008) - | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went - to 't' instead of 'T') ........ - -2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43 - -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make - sure we put greeting files into the database like we do with the - others. Issue #11795 Reported by: dimas Patches: vmgreet.patch - uploaded by dimas (license 88) ........ - -2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix up some T38 state change issues. (closes - issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch - uploaded by dimas (license 88) - - * channels/chan_sip.c: Fix up two scheduling issues. In one - instance a scheduled item was not deleted when it should have - been and in the other it was scheduled again when it shouldn't - have been. - -2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell@digium.com> - - * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge - of this commit was a NoOp, since res_features doesn't exist in - trunk. Thanks to qwell for pointing it out! - - * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) - | 5 lines For some reason, the use of this strdupa() is leading - to memory corruption on freebsd sparc64. This trivial workaround - fixes it. (closes issue #10300, closes issue #11857, reported by - mattias04 and Home-of-the-Brave) ........ - -2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/logger.c: Normalize the detection for execinfo, so that - Linux (glibc) and other platforms with libexecinfo will generate - inline stack backtraces correctly. - -2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell@digium.com> - - * /: Merged revisions 100626 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008) - | 7 lines Fix a crash in ast_masq_park_call() (issue #11342) - Reported by: DEA Patches: res_features-park.txt uploaded by DEA - (license 3) ........ - -2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) | - 1 line Correct a comment which made little/no sense. ........ - -2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell@digium.com> - - * main/channel.c, channels/chan_local.c, /, - include/asterisk/channel.h: Merged revisions 100581 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 - Jan 2008) | 9 lines Make some deadlock related fixes. These bugs - were discovered and reported internally at Digium by Steve Pitts. - - Fix up chan_local to ensure that the channel lock is held - before the local pvt lock. - Don't hold the channel lock when - executing the timing function, as it can cause a deadlock when - using chan_local. This actually changes the code back to be how - it was before the change for issue #10765. But, I added some - other locking that I think will prevent the problem reported - there, as well. ........ - - * main/pbx.c: Clean up some formatting, and simplify a bit of code - using ast_str - -2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't do a network byte order conversion - when setting the socket's port variable to that of bindaddr's. It - is already in the correct network byte order. (closes issue - #11800) Reported by: hmodes - -2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell@digium.com> - - * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert - some spaces to tabs - - * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few - little formatting changes - - * main/channel.c: These readlocks always fail for me on my mac, and - I saw it happen again today on another mac. We ignore the return - value of locking operations almost everywhere in Asterisk. So, - ignore these, as well, so Asterisk will actually work on systems - where this is occurring while I look into what the issue is. - -2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, include/asterisk/sched.h, - channels/chan_iax2.c: With the switch to the ast_sched_replace* - API in trunk, we lose the correction that was just merged from - 1.4, so this is a changeover to those APIs to use the macro - versions, so that we properly detect errors from ast_sched_del, - instead of simply ignoring the return values. - - * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /, - channels/chan_sip.c, channels/chan_h323.c, - include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c, - channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged - revisions 100465 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) - | 11 lines When deleting a task from the scheduler, ignoring the - return value could possibly cause memory to be accessed after it - is freed, which causes all sorts of random memory corruption. - Instead, if a deletion fails, wait a bit and try again (noting - that another thread could change our taskid value). (closes issue - #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt - uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, - stuarth` ........ - -2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker@digium.com> - - * doc/tex/channelvariables.tex: Get rid of that last little bit. - - * build_tools/menuselect-deps.in, configs/vpb.conf.sample - (removed), makeopts.in: Remove more remnants of chan_vpb - -2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson@digium.com> - - * channels/chan_vpb.cc (removed), configure, - include/asterisk/autoconfig.h.in, configure.ac, - channels/Makefile, .cleancount: Removing chan_vpb from the tree - -2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) | - 2 lines This would have never been true, since we're passing - (sizeof(req.data) - 1) as the len to recvfrom(). ........ - -2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_rpt.c: correct a real problem and silence an annoying - compiler warning - -2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Insure that we are not going to pass a NULL - pointer to add_to_interfaces. (closes issue #11840) Reported by: - junky - -2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp@digium.com> - - * main/dial.c, include/asterisk/dial.h: Add an API call that steals - the answered channel so that a destruction of the dialing - structure does not hang it up. - -2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher@digium.com> - - * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as - the default, too - -2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/app.h: ummm... might be good if this macro - argument was actually used :-) - - * include/asterisk/app.h: add the ability to define a structure - type for argument parsing when it would be useful to be able to - pass it between functions - -2008-01-24 22:02 +0000 [r100266] James Golovich <james@gnuinter.net> - - * channels/chan_sip.c: Fix simple whitespace issue - -2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24 - Jan 2008) | 2 lines make these macros not assume that the only - other field in the structure is 'argc'... this is true when - someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable - to define your own structure as long as it has the right fields - ........ - -2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp@digium.com> - - * main/features.c: Minor cosmetic change... - -2008-01-24 18:35 +0000 [r100224] James Golovich <james@gnuinter.net> - - * main/astmm.c: Increase the size of filenames stored when astmm is - used. If the path length was long they would be truncated and - grouped together with whatever matches - -2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp@digium.com> - - * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp - branch. This adds a strictrtp option to rtp.conf which drops - packets that do not come from the remote party. (closes issue - #8952) Reported by: amorsen - -2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: Merged revisions 100164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008) - | 2 lines Update main Asterisk copyright info to 2008 ........ - -2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker@digium.com> - - * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) | - 6 lines Fix compilation on Solaris. (closes issue #11832) - Patches: bug-11832.diff uploaded by snuffy (license 35) ........ - - * channels/chan_sip.c, main/features.c: Move chan_local dependency - into places (only one) that previously depended on res_features, - and used local channels - -2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, - channels/chan_mgcp.c: Remove dependency on res_features from some - channel drivers. It is now part of the core and no longer exists - as a module. - - * main/channel.c: Some more cosmetic changes. - - * main/channel.c: Add some spacing. - - * main/dial.c: Test hopefully over. - - * main/dial.c: Testing something... - -2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix flag bit definitions to make code from - issue #11049 actually work; along the way, clarify comments and - add some dummy flag definitions for other multi-bit flags to - hopefully stop this from happening in the future (closes issue - #11049) - -2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker@digium.com> - - * res/res_features.c (removed), main/Makefile, main/features.c - (added), include/asterisk/_private.h, CHANGES, .cleancount, - main/asterisk.c, main/loader.c, include/asterisk/features.h: Move - code from res_features into (new file) main/features.c - -2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell@digium.com> - - * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your - contributions! - -2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7 - lines Second attempt. Don't change invitestate when receiving 18x - messages in CANCEL state. (issue #11736) Reported by: MVF Patch - by oej. ........ - - * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 - lines Make sure we don't cancel destruction on calls in CANCEL - state, even if we get 183 while waiting for answer on our CANCEL. - (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by - oej (license 306) Tested by: MVF ........ - -2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23 - Jan 2008) | 3 lines Fixing a typo. ........ - -2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | - 8 lines ChanSpy issues a beep when it starts at the beginning of - a list of channels to potentially spy on. However, if there were - no matching channels, it would beep at you over and over, which - is pretty annoying. Now, it will only beep once in the case that - there are no channels to spy on, but it will still beep again - once it reaches the beginning of the channel list again. (closes - issue #11738, patched by me) ........ - - * main/tcptls.c: Fix tcptls build when openssl isn't installed - (closes issue #11813) Reported by: tzafrir Patches: - asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176) - -2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: since echo canceler parameters in Zaptel are - now signed integers, allow them during parsing - -2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_h323.c: Progress messages don't work (closes issue - #10497) Reported by: pj Patches: h323-announces-r99483.diff - uploaded by sergee (license 138) Tested by: pj - -2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure - that we are aware of a pending INVITE even if we're using TCP - -2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Coding guidelines fixups - - * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) - | 8 lines When we reset the password via an external command, we - should also reset the password stored in the in-memory list, too - (otherwise it doesn't really take effect). (closes issue #11809) - Reported by: davetroy Patches: fix_externpass.diff uploaded by - davetroy (license 384) ........ - - * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008) - | 2 lines Oops, should have checked for a NULL obj, here, too - ........ - - * res/res_config_ldap.c: Coding guidelines cleanup - - * /, main/acl.c: Merged revisions 99718 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008) - | 2 lines Just confirmed that all current platforms need this - header file ........ - - * /: Oops - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, doc/ldap.txt (added), - configure.ac, configs/res_ldap.conf.sample (added), - res/res_config_ldap.c (added), CHANGES, makeopts.in, - contrib/scripts/asterisk.ldap-schema (added), - contrib/scripts/asterisk.ldif (added): Add res_config_ldap for - realtime LDAP engine. (closes issue #5768) Reported by: mguesdon - Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon - (license 121) res_ldap.conf.sample uploaded by suretec (license - 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) - asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested - by: oej, mguesdon, suretec, cthorner - -2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 - lines Thanks to Russell's education I realize that BUFSIZ has - changed since I learned the C language over 20 years ago... - Resetting chan_sip to the size of BUFSIZ that I expected in my - old head to avoid too heavy memory allocations on some systems. - ........ - - * doc/tex/channelvariables.tex, CHANGES: Documentation updates for - BRIDGEPVTCALLID - -2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher@digium.com> - - * /, main/acl.c: Merged revisions 99643 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008) - | 2 lines Fix the defines for OS X (and Solaris, too) ........ - -2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Make sure the command is not just present but is - also configured to be executed - -2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej@edvina.net> - - * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: - Add a generic function to set the bridged call PVT unique id - string as a channel variable BRIDGEPVTCALLID This is important - for call tracing in log files and CDRs, so that the SIP callID - can be traced along servers. The CHANNEL dialplan function won't - work here, since the outbound channel is gone when we need the - Call-ID. Other channel drivers may now implement the same - function :-), but this patch only supports chan_sip.so. Inspired - by (issue #11816) Reported by: ctooley Patch by oej - -2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell@digium.com> - - * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change - the Asterisk CLI startup commands feature to read commands to run - from cli.conf after a discussion on the -dev list. - -2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej@edvina.net> - - * channels/chan_local.c, /, res/res_features.c, - channels/chan_agent.c, apps/app_followme.c: Merged revisions - 99594 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 - lines Add more dependencies on chan_local and add a note to the - description of chan_local so that people don't disable it in - menuselect just to clean up. ........ - - * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 - lines Add dependency on chan_local to app_dial. Dial still runs - without chan_local, but will be missing forwarding functionality. - ........ - -2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher@digium.com> - - * /, main/acl.c: Merged revisions 99540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) - | 7 lines Ensure that we can get an address even when we don't - have a default route. (closes issue #9225) Reported by: junky - Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 - (license 14) Tested by: oej, loloski, sergee ........ - -2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Point out a bug in some debug counter - handling - -2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add authentication options to the SIP - dialstring. Documentation follows separately (issue #11587) - Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by - sobomax (license 359) - - * configs/sip.conf.sample: Documentation updates - - * doc/siptls.txt: Small fixes - - * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c, - include/asterisk/tcptls.h: Doxygen updates - -2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: Merged revisions 99426 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 - Jan 2008) | 12 lines Fixing an issue wherein monitoring local - channels was not possible. During a channel masquerade, the - monitors on the two channels involved are swapped. In 99% of the - cases this results in the desired effect. However, if monitoring - a local channel, this caused the monitor which was on the local - channel to get moved onto a channel which is immediately hung up - after the masquerade has completed. By swapping the monitors - prior to the masquerade, we avoid the problem by tricking the - masquerade into placing the monitor back onto the channel where - we want it. During the investigation of the issue, the channel's - monitor was the only thing that was swapped in such a manner - which did not make sense to have done. All other variable - swapping made sense. ........ - -2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Fix distinctive ring detection. Reported by: - milazzo Patches: drings.diff uploaded by milazzo (license 383) - Closes issue #11799 - -2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: Adding the - QUEUENAME variable to the variables set using the setqueuevar - option in queues.conf. Suggestion comes from Shaun2222 on IRC. - -2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej@edvina.net> - - * channels/chan_console.c: Remove compiler warning for - uninitialized variable - - * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was - committed without any doxygen obviously. Tss tss. - - * channels/chan_sip.c: Updating doxygen - -2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, /, res/res_odbc.c, - configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 - Jan 2008) | 8 lines Permit the user to specify number of seconds - that a connection may remain idle, which fixes a crash on - reconnect with the MyODBC driver. (closes issue #11798) Reported - by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt - uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ - -2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 - lines Bump the buffer size for Via headers up to 512. There are - some exceptionally large Via headers out there. (closes issue - #11783) Reported by: ofirroval ........ - -2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej@edvina.net> - - * CREDITS: Update - -2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Change over to using ast_debug so these - debug messages don't always show up. - -2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell@digium.com> - - * channels/chan_console.c: Add a "console active" CLI command, - which lets you find out which console device is currently active - for the Asterisk CLI, or to set it. Also, knock multiple device - support off of the to-do list. - - * configs/console.conf.sample: correct the name of a CLI command - for getting available device names - - * configs/console.conf.sample, channels/chan_console.c: Merge - changes from team/russell/console_devices - Add support for - multiple devices. All devices are configured in console.conf. - - Add "console list devices" CLI command to show configured - devices. Also, changed the old "list devices" to be "list - available", which queries PortAudio for all audio devices that - are available for use. - - * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | - 4 lines Fix a couple of memory leaks with frame handling. - Specifically, ast_frame_free() needed to be called on the frame - that came from the translator to signed linear. ........ - - * README: Add Cygwin as an "other" platform that is supported - - * README: Various README updates - -2008-01-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.6.0-beta1 released. - -2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell@digium.com> - - * CREDITS, include/asterisk/http.h, main/tcptls.c (added), - main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), - main/Makefile, main/http.c, include/asterisk/tcptls.h (added), - configs/sip.conf.sample, CHANGES: Merge changes from - team/group/sip-tcptls This set of changes introduces TCP and TLS - support for chan_sip. There are various new options in - configs/sip.conf.sample that are used to enable these features. - Also, there is a document, doc/siptls.txt that describes some - things in more detail. This code was implemented by Brett Bryant - and James Golovich. It was reviewed by Joshua Colp and myself. A - number of other people participated in the testing of this code, - but since it was done outside of the bug tracker, I do not have - their names. If you were one of them, thanks a lot for the help! - (closes issue #4903, but with completely different code that what - exists there.) - - * main/frame.c, /, include/asterisk/translate.h: Merged revisions - 99081 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | - 9 lines Revert adding the packed attribute, as it really doesn't - make sense why that would do any good. Fix the real bug, which is - to do the check to see if the frame came from a translator at the - beginning of ast_frame_free(), instead of at the end. This - ensures that it always gets checked, even if none of the parts of - the frame are malloc'd, and also ensures that we aren't looking - at free'd memory in the case that it is a malloc'd frame. (closes - issue #11792, reported by explidous, patched by me) ........ - - * /, include/asterisk/translate.h: Merged revisions 99079 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | - 4 lines Since we're relying on the offset between the frame and - the beginning of the translator pvt struct, set the packed - attribute to make sure we get to the right place. (potential fix - for issue #11792) ........ - -2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson@digium.com> - - * res/res_features.c: This should at least temporarily fix a - problem where the 't' Dial option is incorrectly passed to the - transferee when built-in attended transfers are used. There is - still a problem with 'T', but better to fix some problems than no - problems while we work on it. (closes issue #7904) Reported by: - k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee - (license 138) Tested by: sergee, otherwiseguy - -2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for - speed (closes issue #10723) Reported by: mnicholson Patches: - func-odbc-direct-execute1.diff uploaded by mnicholson (license - 96) Tested by: Corydon76, mnicholson, falves11 - - * res/res_odbc.c: Permit username and password to be NULL (which - enables pass-through from the layer above). Reported by: lurcher - Patch by: tilghman (Closes issue #11739) - - * funcs/func_cut.c: Reset default CUT delimiter back to '-' - -2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell@digium.com> - - * channels/chan_console.c: Make the output of "console list - devices" a bit prettier. - - * channels/chan_console.c: List which devices are inputs and - outputs in "console list devices" - - * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for - ast_best_codec() - - * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h: - Merged revisions 99004 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | - 10 lines Have IAX2 optimize the codec translation path just like - chan_sip does it. If the caller's codec is in our codec list, - move it to the top to avoid transcoding. (closes issue #10500) - Reported by: stevedavies Patches: iax-prefer-current-codec.patch - uploaded by stevedavies (license 184) - iax-prefer-current-codec.1.4.patch uploaded by stevedavies - (license 184) Tested by: stevedavies, pj, sheldonh ........ - -2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue - #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded - by DEA (license 3) - -2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker@digium.com> - - * Makefile, build_tools/cflags.xml, channels/chan_zap.c, - main/dsp.c, configs/zapata.conf.sample: Add several busy - detection related defines to menuselect. Allow better busy detect - debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and - BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: - busydetect_enhancement.patch uploaded by agx (license 298) - busydetect-r94975.diff uploaded by sergee (license 138) - Additional changes/cleanup by me. - -2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: state_interface could be NULL, so use the - never-NULL cur->state_interface for this check - - * apps/app_queue.c: Get the device state of the state interface - instead of the interface when creating a new queue member. Thanks - to Atis Lezdins for bringing this up on the Asterisk-Dev mailing - list. - -2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker@digium.com> - - * /, configs/zapata.conf.sample: Merged revisions 98991 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600 - (Thu, 17 Jan 2008) | 4 lines Add a clarification about the - immediate= option of zapata.conf Issue 11784, patch by klaus3000. - ........ - -2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: major - reliability and performance improvement in VWMI monitoring for - FXO ports (code by markster, me and dbailey) - - * res/res_config_curl.c: resolve (valid) compiler warning about - variable that could be used before being initialized - -2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c, doc/tex/phoneprov.tex, - configs/phoneprov.conf.sample: Update res_phoneprov to default to - setting the SERVER variable to the IP the HTTP request for the - config came in on and the SERVER_PORT to the bindport setting in - sip.conf. I've left in the ability to override these options, - because I can't always guess how someone might decide to do - something weird with what is available to them--although needing - to is pretty unlikely. Documentation was updated to reflect - preference for not setting serveraddr, serveriface, or - serverport. Tested on Linux and OS X. - -2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature - works, by allowing it to be a column NOT logged into the - database. This will allow more granularity of a decision - evaluated in the dialplan, then takes effect when posting the - CDR. - -2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell@digium.com> - - * CHANGES, main/asterisk.c: Add support for an easy way to - automatically execute some Asterisk CLI commands immediately at - startup. Any commands in the startup_commands file in the - Asterisk config diretory will get executed. (closes issue #11781) - Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt - uploaded by jamesgolovich (license 176) -- With some changes by - me. - -2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build - against <package>_LIB, per recommendations from Russell. - -2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: Info about res_config_curl - -2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_curl.c (added), main/utils.c: New module - res_config_curl (closes issue #11747) Reported by: Corydon76 - Patches: res_config_curl.c uploaded by Corydon76 (license 14) - 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14) - Tested by: jmls - -2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell@digium.com> - - * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the - changes from issue #10665 from the team/group/sip_session_timers - branch. This set of changes introduces SIP session timers support - (RFC 4028). In short, this prevents stuck SIP sessions that were - not properly torn down due to network or endpoint failures during - an established SIP session. To quote some of the documentation - supplied with the patch: "The SIP Session-Timers is an extension - of the SIP protocol that allows end-points and proxies to refresh - a session periodically. The sessions are kept alive by sending a - RE-INVITE or UPDATE request at a negotiated interval. If a - session refresh fails then all the entities that support Session- - Timers clear their internal session state. In addition, UAs - generate a BYE request in order to clear the state in the proxies - and the remote UA (this is done for the benefit of SIP entities - in the path that do not support Session-Timers)." (closes issue - #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by - rjain (license 226) chan_sip.c.diff uploaded by rjain (license - 226) sip.conf.sample.diff uploaded by rjain (license 226) - proc_422_rsp_comment.diff uploaded by rjain (license 226) - chan_sip.c.cache.diff uploaded by rjain (license 226) - chan_sip.memalloc uploaded by rjain (license 226) - chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches - tracked in team/group/sip_session_timers, with some additional - fixes by russell and oej. Tested by: jtodd, rjain, loloski - -2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - Partially revert r93898, because it broke the way netsnmp was - being detected. rizzo, do you want to discuss so we can rethink - this, or do you have another way? - - * CHANGES: Add note about new update.log to CHANGES, by request of - jmls and further prodding by jsmith. - - * Makefile, /: Add logging for 'make update' command (also fixes - updates in some places). Issue #11766, initial patch by jmls. - -2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 - lines Add missing NULLs at end of two ast_load_realtimes. (closes - issue #11769) Reported by: tequ Patches: chaniax.patch uploaded - by dimas (license 88) ........ - -2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: Merged revisions 98964 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 - Jan 2008) | 10 lines Fix a deadlock in chan_local in - local_hangup. There was contention because the local_pvt was held - and it was attempting to lock a channel, which is the incorrect - locking order. (closes issue #11730) Reported by: UDI-Doug - Patches: 11730.patch uploaded by putnopvut (license 60) Tested - by: UDI-Doug ........ - -2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c: Make users list static - -2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp@digium.com> - - * main/dial.c, /: Merged revisions 98960 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 - lines Introduce a lock into the dialing API that protects it when - destroying the structure. (closes issue #11687) Reported by: - callguy Patches: 11687.diff uploaded by file (license 11) - ........ - - * /, main/rtp.c: Merged revisions 98958 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 - lines Add two more SDP names for ulaw and alaw. (closes issue - #11777) Reported by: tootai ........ - - * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 - lines Don't drop the old record route information when dealing - with packets related to a reinvite. (closes issue #11545) - Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by - kebl0155 (license 356) ........ - - * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This - seems to come from way back when and I can't think of a reason - for it being here, plus it could cause needless DNS lookups. - (closes issue #10983) Reported by: jtodd - -2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf@digium.com> - - * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found - this problem with running the expr2 parser on OSX. Make the - #defines come out the same between the parser & lexer. - -2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp@digium.com> - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, codecs/codec_speex.c, - configure.ac, makeopts.in: Merged revisions 98951 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan - 2008) | 4 lines Add autoconf logic for speexdsp. Later versions - use a separate library for some things so we need to use it if - present in codec_speex. (closes issue #11693) Reported by: yzg - ........ - -2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | - 11 lines Change a buffer in check_auth() to be a thread local - dynamically allocated buffer, instead of a massive buffer on the - stack. This fixes a crash reported by Qwell due to running out of - stack space when building with LOW_MEMORY defined. On a very - related note, the usage of BUFSIZ in various places in chan_sip - is arbitrary and careless. BUFSIZ is a system specific define. On - my machine, it is 8192, but by definition (according to google) - could be as small as 256. So, this buffer in check_auth was 16 - kB. We don't even support SIP messages larger than 4 kB! Further - usage of this define should be avoided, unless it is used in the - proper context. ........ - -2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: - Add the "filter" keyword - -2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell@digium.com> - - * main/translate.c, include/asterisk/translate.h: Clean up - something I did for ABI compatability in 1.4 - - * main/frame.c, /, main/translate.c, main/abstract_jb.c, - channels/chan_iax2.c, codecs/codec_zap.c, - include/asterisk/frame.h, main/rtp.c, - include/asterisk/translate.h: Merged revisions 98943 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 - Jan 2008) | 25 lines Commit a fix for some memory access errors - pointed out by the valgrind2.txt output on issue #11698. The - issue here is that it is possible for an instance of a translator - to get destroyed while the frame allocated as a part of the - translator is still being processed. Specifically, this is - possible anywhere between a call to ast_read() and - ast_frame_free(), which is _a lot_ of places in the code. The - reason this happens is that the channel might get masqueraded - during this time. During a masquerade, existing translation paths - get destroyed. So, this patch fixes the issue in an API and ABI - compatible way. (This one is for you, paravoid!) It changes an - int in ast_frame to be used as flag bits. The 1 bit is still used - to indicate that the frame contains timing information. Also, a - second flag has been added to indicate that the frame came from a - translator. When a frame with this flag gets released and has - this flag, a function is called in translate.c to let it know - that this frame is doing being processed. At this point, the flag - gets cleared. Also, if the translator was requested to be - destroyed while its internal frame still had this flag set, its - destruction has been deffered until it finds out that the frame - is no longer being processed. Admittedly, this feels like a hack. - But, it does fix the issue, and I was not able to think of a - better solution ... ........ - -2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 - lines Based on the boundary found move over the correct amount. - (closes issue #11750) Reported by: tasker ........ - - * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 - lines Accept "; boundary=" not just ";boundary=" in the multipart - mixed content type. (closes issue #11750) Reported by: tasker - ........ - -2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add - backupdeleted option to app_voicemail (closes issue #10740) - Reported by: ruffle Patches: app_voicemail.diff uploaded by - ruffle (license 201) 10740-voicemail.diff uploaded by qwell - (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak - (license 7) Tested by: blitzrage, mvanbaak, qwell - -2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson@digium.com> - - * apps/app_directory.c: Big improvement for app_directory. This - patch breaks the do_directory function up so that it is more - easily parsed by the human brain. It also fixes some errors. I'll - quote dimas from the original bug description: "app_directory - contained some duplicate code even before addition of 'm' option. - Addition of that option doubled amount of that code. Worst of - all, there are minor differences between these code block and - bugs caused by these differences. 1. There is a memory leak. In - the 'menu' mode, result of the convert(pos) function is not freed - while it should be. 2. In the 'menu' mode check for - OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, - application works in the mode opposite to what user expect - (checking last name when user wants the first nd vice versa). 3. - select_item function plays message for user using res = func1() - || func2() || func3()... construct. This construct loses the - actual value returned by ast_waitstream() for example so at the - end, res does not contain digit user dialed while listening to - the message. 4. (also in 1.4) application announces entries from - voicemail.conf/realtime separately from entries from users.conf. - I see no reason why doing so instead of building combined list. - 5. Alot of duplicated code as already mentioned." This was tested - by dimas and I (I tested under valgrind). A word of caution: any - bug fixes that happen in app_directory in 1.4 will almost - certainly not merge cleanly into trunk as a result of this, but - it is well worth it. Huge thanks to dimas for this wonderful - submission. (closes issue #11744) Reported by: dimas Patches: - dir3.patch uploaded by dimas (license 88) Tested by: putnopvut, - dimas - -2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Make sure the user's manager secret exists, even - if it is blank. (closes issue #11749) Reported by: srt - -2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson@digium.com> - - * CHANGES: Add description of TOUPPER and TOLOWER dialplan - functions to CHANGES. - -2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Add proper call forwarding (all and busy) - support for chan_skinny. Note: NoAnswer support is currently not - implemented, as it would take a significant amount of work to - figure out how to do correctly. Closes issue #11310, patches, - testing, and support by DEA, mvanbaak, and myself. - -2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell@digium.com> - - * /, main/translate.c: Merged revisions 98774 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | - 3 lines Revert a change that introduces an unacceptable - performance hit and is causing memory leaks ... (from rev 97973) - ........ - -2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix for potential crash with vmexten - -2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Merged revisions 98737 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan - 2008) | 3 lines Fixing another compilation error. I'm a bit off - today :( ........ - - * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan - 2008) | 8 lines Adding explicit defaults for missing options to - init_queue. This is necessary because if a user either removes or - comments one of these options and reloads their queues, the - option will not reset to its default, instead maintaining the - value from prior to the reload. Thanks to John Bigelow for - pointing this error out to me. ........ - -2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Print out a warning when spaces are used in the - variable name in Set and MSet. It is extremely hard to debug this - issue so this should make it easier. (closes issue #11759) - Reported by: caio1982 Patches: setvar_space_warning1.diff - uploaded by caio1982 (license 22) - - * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update - documentation. (closes issue #11763) Reported by: IgorG Patches: - docupd.v1.diff uploaded by IgorG (license 20) - -2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell@digium.com> - - * apps/app_jack.c: Add another small option for the JACK app and - JACK_HOOK function. The 'n' option tells JACK not to start jackd - automatically if it is not already running. Otherwise, the - default is that jackd will get started for you if it isn't - running already. - - * CHANGES: - Break up the Misc. section a bit with a new section - for Misc. New Modules - Change spacing a bit in some places for - consistent indentation - - * CHANGES, apps/app_jack.c (added): Bring in the code from - team/russell/jack/. Add a new module, app_jack, which provides - interfaces to JACK, the Jack Audio Connection Kit - (http://www.jackaudio.org/). Two interfaces are provided; there - is a JACK() application, and a JACK_HOOK() function. Both - interfaces create an input and output JACK port. The application - makes these ports the endpoint of the call. The audio coming from - the channel goes out the output port and whatever comes back in - on the input port is what gets sent to the channel. The - JACK_HOOK() function turns on a JACK audiohook on the channel. - This lets you run the audio coming from a channel through JACK, - and whatever comes back in is what gets forwarded on as the - channel's audio. This is very useful for building custom vocoders - or doing recording or analysis of the channel's audio in another - application. In case anyone is curious, the platform that - inspired me to write this is PureData (http://puredata.info/). I - wrote these JACK interfaces so that I could use Pd to do - interesting things with the audio of phone calls ... - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add - configure script check for JACK. - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: - Remove KDE configure script check that isn't used - - * main/audiohook.c: Remove a duplicate lock of the audiohook lock - when destroying manipulate audiohooks. This causes an error when - we attempt to destroy the lock later when freeing the audiohook. - - * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar", - which allows you to set a channel variable (or function) on an - active channel from the CLI. - -2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: Conversion to load manager.conf into memory did - not convert the password functions correctly. (Closes issue - #11749) - -2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand@digium.com> - - * /, main/http.c: merging a comment added in 1.4 - -2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and - ensure that the AMI DNDState event always gets generated. (closes - issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded - by tzafrir (modified by me) (license 46) - -2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) - | 4 lines Add a connection timeout attribute, as that was what - was intended with the login timeout, but ODBC divides it up into - 2 different timeouts. (Closes issue #11745) ........ - -2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell@digium.com> - - * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some - extra checking to help out with a potential error when trying to - run "make asterisk.pdf" when not all of the right packages are - installed. (closes issue #10763) Reported by: Corydon76 Patches: - 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76 - -2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add - 'auto' signalling mode for Zaptel channels. (closes issue #11690) - Reported by: tzafrir Patches: signaling_to_signalling.diff - uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded - by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir - (license 46) zap_no_default_sig.diff uploaded by tzafrir (license - 46) zap_signal_auto.diff uploaded by tzafrir (license 46) - -2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp@digium.com> - - * main/event.c: Goodbye again drumkilla. - - * main/event.c: drumkilla ftw. - - * main/audiohook.c: I am no longer Rockin' - - * main/audiohook.c: Testing something... - -2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | - 9 lines Fix up setting the EID on BSD based systems. (closes - issue #11646) Reported by: caio1982 Patches: - dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) - dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested - by: caio1982, mvanbaak ........ - -2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 98325 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 - lines If the incoming RTP stream changes codec force the bridge - to break if the other side does not support it. (closes issue - #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch - uploaded by tsearle (license 373) ........ - - * /, res/res_agi.c: Merged revisions 98317 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 - lines If the channel is hungup during RECORD FILE send a result - code of -1 to be uniform with everything else. (closes issue - #11743) Reported by: davevg Patches: res_agi.diff uploaded by - davevg (license 209) ........ - -2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 98315 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan - 2008) | 5 lines Properly report the hangup cause as no answer - when someone does not answer (closes issue #10574, reported by - boch, patched by moy) ........ - -2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell@digium.com> - - * codecs/codec_resample.c: Kevin noted that the thing that I - _actually_ changed here was that I converted a value from a - double, to a float, back to a double. Sure enough, when I changed - my interim variable back to a double, it still blows up. - Switching all of these to a float fixes the problem. This seems - like a compiler bug where a double passed as an argument isn't - getting properly aligned, so I'll have to see if I can replicate - it with a small test program. (related to issue #11725) - - * codecs/codec_resample.c: Fix a bus error that happened when - asterisk was built with optimizations on with platforms that - explode on unaligned access. I'm not exactly sure why this fixes - it, but it fixed it on the machine I was testing on. If it makes - sense to you, feel free to enlighten me. :) (closes issue #11725, - patched by me) - -2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to - adaptive_odbc, too - - * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix - -2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp@digium.com> - - * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 - lines Ensure the return value of ast_bridge_call is passed back - up as the application return value. This is needed for transfers - to function so the PBX core knows to continue execution. (closes - issue #10327) Reported by: kkiely ........ - -2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell@digium.com> - - * codecs/codec_g722.c: At one point during working on this module, - I had the lin/lin16 versions of the framein callbacks different. - However, they are now the same again, so remove the duplicate - code and use the same functions for the lin/lin16 versions. - -2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) - | 2 lines Back out changes from revision 97077, since it wasn't - perfect ........ - - * doc/manager_1_1.txt: Documentation updates - -2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO - using '!' as the 'digit' - -2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell@digium.com> - - * codecs/codec_g722.c, main/frame.c: - Fix the last set of places - where incorrect assumptions were made about the sample length - with g722. It is _2_ samples per byte, not 1. This was all over - the place, and I believed it, and it is what caused me to take so - long to figure out what was broken. - Update copyright - information on codec_g722. - -2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson@digium.com> - - * main/translate.c: Fix "core show translation" to not output - information for "unknown" codecs. This fix was made in favor of - the proposed patch since it doesn't involve changing a core codec - define. (closes issue #11722, reported and initially patched by - caio1982, final patch by me) - -2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new - global and per-peer option to chan_sip, qualifyfreq, which allows - you to set the qualify frequency. (closes issue #11597) Reported - by: wilder Patches: qualifyfreq5.patch uploaded by wilder - (license 362) -- with some mods by me - - * main/translate.c: Simplify this code with a suggestion from Luigi - on the asterisk-dev list. Instead of using is16kHz(), implement a - format_rate() function. - -2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) - | 6 lines 1) When we get a translated frame out, clone it, - because if the translator pvt is freed before we use the frame, - bad things happen. 2) Getting a failure from ast_sched_delete - means that the schedule ID is currently running. Don't just - ignore it. (Closes issue #11698) ........ - -2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant <russell@digium.com> - - * /, main/translate.c: Merged revisions 97976 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | - 3 lines Fix various timing calculations that made assumptions - that the audio being processed was at a sample rate of 8 kHz. - ........ - - * codecs/codec_g722.c: Fix various issues in codec_g722. - The most - common fix being made here is to fix all of the places where the - number of output samples and output bytes gets updated in the - translator state structure. - Fix a number of other places where - the number of samples provided as an initialization value to a - struct was incorrect. - - * codecs/codec_resample.c: Fix the buffer_samples value. For signed - linear, the number of samples needed to fill the buffer is half - the buffer size. - -2008-01-10 21:58 +0000 [r97933] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan - 2008) | 6 lines Let us leave a voicemail for ourself if we have - logged into VoiceMailMain and chosen to leave a message. (closes - issue #11735, reported and patched by jamessan) ........ - -2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy <murf@digium.com> - - * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged - revisions 97889 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 - line Applied the same fixes for ael.flex as was done in 97849 for - ast_expr2.fl; overrode the normally generate yyfree func with our - own version that checks the pointer for non-null before passing - to free(). Also takes care of a little problem with 2.5.33 and - the use of the __STDC_VERSION__ macro. ........ - - * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged - revisions 97849 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 - line This is a fix for 2 things: a problem Terry was having in - OSX with null pointers, which was my fault, as I probably forgot - to run the sed script last time I made mods. So, I moved the fix - into the flex input itself. Then, I found when I used flex - 2.5.33, that it was using __STDC_VERSION__, and that's not real - good; so I added back in a DIFFERENT sed script to fix that - little mess. Tested everything, a couple different ways. Hope I - did no harm, at the least. ........ - -2008-01-10 20:13 +0000 [r97848] Jason Parker <jparker@digium.com> - - * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan - 2008) | 1 line Fix a comment that is no longer true. ........ - -2008-01-10 20:05 +0000 [r97846] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Use the appropriate line ending for the - X-Asterisk-VM-Message-Type header. (closes issue #11734, reported - and patched by jaroth) - -2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson <twilson@digium.com> - - * main/ast_expr2f.c: heh, remove patch to generated file. - - * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was - getting WARNINGS under MALLOC_DEBUG) - -2008-01-10 17:38 +0000 [r97805] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_odbc.c: Fix problem with timestr going out of scope - (Closes issue #11726, closes issue #11731) - -2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant <russell@digium.com> - - * formats/format_sln16.c: minor formatting changes - - * main/translate.c: spaces to tabs - - * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK - check again. I changed this to an inline implementation to fix a - small bug, but after a discussion with rizzo, I went to change it - back. Also, it turns out that the implementation of the macro - already supported what was needed to fix the problem. - - * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample, - pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | - 2 lines Remove other remnants of pbx_kdeconsole ........ - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, - pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | - 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in - ages, and nobody has complained. (closes issue #11706, reported - by caio1982) ........ - -2008-01-10 15:12 +0000 [r97698] Joshua Colp <jcolp@digium.com> - - * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan - 2008) | 6 lines Don't try to copy the category from the group if - no category exists. (closes issue #11724) Reported by: IgorG - Patches: group_count.v1.patch uploaded by IgorG (license 20) - ........ - -2008-01-10 00:54 +0000 [r97657] Russell Bryant <russell@digium.com> - - * include/asterisk.h: These prototypes are not supposed to be in - asterisk.h. They are already in version.h. - -2008-01-10 00:50 +0000 [r97656] Steve Murphy <murf@digium.com> - - * include/asterisk.h, channels/console_video.c, utils/astman.c, - channels/console_board.c, channels/vgrabbers.c: The fixes in this - commit are mainly to allow compiling of trunk with - --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, - the version.c needs to be in the OBJS line; asterisk.h was chosen - to have the prototypes for ast_get_version, ast_get_version_num; - and the ASTERISK_FILE_VERSION macro needs to be used after - including asterisk.h in a few files. I hope I did the right - thing. If not, let me know. - -2008-01-10 00:39 +0000 [r97655] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: oops, missed the case of a 0 permission (which - should mean everybody is allowed, not nobody) - -2008-01-10 00:22 +0000 [r97653] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c: Attempt at making lookup_iface work under - FreeBSD. Not yet tested, but it compiles under OS X. And still - works under linux. - -2008-01-10 00:17 +0000 [r97652] Russell Bryant <russell@digium.com> - - * codecs/Makefile: Fix this so it doesn't force codec_g722 to get - relinked every time - -2008-01-10 00:12 +0000 [r97651] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, main/manager.c, channels/chan_sip.c, - res/res_features.c, pbx/pbx_realtime.c, - configs/manager.conf.sample, CHANGES, channels/chan_iax2.c, - include/asterisk/manager.h, apps/app_stack.c, main/db.c, - apps/app_voicemail.c: Several manager changes: 1) Add the - Dialplan class, for NewExten and VarSet events, which should cut - down on the volume of traffic in the Call class. 2) Permit some - commands to be run from multiple classes, such as allowing DBGet - to be run from either the System or the Reporting class. 3) - Heavily document each class in the sample config, as there were - several that made no sense to be in the write= line, and two that - made no sense to be in the read= line (since they controlled no - permissions there). (Closes issue #10386) - -2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant <russell@digium.com> - - * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the - files changes - - * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | - 2 lines Strip terminal sequences from the verbose messages - ........ - - * configure: re-gen configure - - * configure.ac: re-add check for gtk1, which is used for - pbx_gtkconsole (related to issue #11706) - - * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | - 3 lines Make pbx_gtkconsole build ... but doesn't actually load - on my system still (related to issue #11706) ........ - -2008-01-09 21:37 +0000 [r97634] Terry Wilson <twilson@digium.com> - - * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml, - phoneprov/polycom.xml, res/res_phoneprov.c (added), - funcs/func_strings.c, phoneprov/000000000000-phone.cfg, - configs/modules.conf.sample, main/acl.c, - include/asterisk/localtime.h, CHANGES, - configs/phoneprov.conf.sample (added), Makefile, phoneprov - (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c, - doc/tex/asterisk.tex: Added a new module, res_phoneprov, which - allows auto-provisioning of phones based on configuration - templates that use Asterisk dialplan function and variable - substitution. It should be possible to create phone profiles and - templates that work for the majority of phones provisioned over - http. It is currently only intended to provision a single user - account per phone. An example profile and set of templates for - Polycom phones is provided. NOTE: Polycom firmware is not - included, but should be placed in AST_DATA_DIR/phoneprov/configs - to match up with the included templates. - -2008-01-09 20:30 +0000 [r97620-97623] Jason Parker <jparker@digium.com> - - * /, main/cli.c: Merged revisions 97622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600 - (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a - command could not be found. Also fix a comment which may have led - to this happening. Issue 11718, reported by kshumard. ........ - - * /, main/cli.c: Merged revisions 97618 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 - line Fix some locking and return value funkiness. We really - shouldn't be unlocking this lock inside of a function, unless we - locked it there too. ........ - -2008-01-09 18:53 +0000 [r97577] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan - 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some - smaller functions this time ........ - -2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c: remove a wrong 'const' - - * images/kpad2.jpg: add annotations for the two message windows we - use. - -2008-01-09 18:04 +0000 [r97531] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 97529 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | - 2 lines Fix saying the parking space number to the caller doing - the parking ... ........ - -2008-01-09 18:03 +0000 [r97530] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/console_board.c, - channels/console_video.h: Two changes: - support scrolling of - message window; - simplify the code for creating a message - window, and try it using a second one in the top of the keypad - (where we echo the dialed number). The 'skin' that supports these - two windows will be committed separately. - -2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming <kpfleming@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) - | 2 lines report the same message whether Zaptel does not have - transcoder support loaded or no transcoders were found ........ - -2008-01-09 16:59 +0000 [r97490] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 - Jan 2008) | 7 lines Set the caller id within the gtalk_alloc - function. As underlined in issue #10437 by Josh, we need to - prevent a possible memory leak. We only set the name part of the - caller id, the number part is not relevant when dealing with - JIDs. Closes issue #11549. ........ - -2008-01-09 16:44 +0000 [r97488] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/console_video.c, - channels/console_board.c, channels/console_video.h: Implement - keyboard handling, and use it to enter a number to dial in the - 'message' area under the keypad. Now you can make calls using the - keypad as a regular phone (or the keyboard for chars not present - on the keypad) - -2008-01-09 16:13 +0000 [r97451] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6 - lines Don't do conferencing totally in Zaptel if Monitor is - running on the channel. (closes issue #11709) Reported by: - BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license - 371) ........ - -2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) - | 2 lines pass the right variable to get an error string... oops - ........ - - * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) - | 2 lines add error number output to ioctl failure messages to - help with debugging ........ - -2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo <rizzo@icir.org> - - * channels/console_video.c, channels/console_video.h: implement the - "console startgui" and "console stopgui" commands so you can - start and stop the gui even outside of a call. This is convenient - for testing, and also for using the keypad to pick up a call, and - to dial a number (the latter not yet implemented, but should be - close). - - * channels/chan_oss.c: make get_video_desc() return the active - console if passed a null argument (channel). - -2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: New option in trunk, needs strdupa to be safe, - too - - * /, main/editline/readline.c, main/cli.c: Merged revisions 97350 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) - | 5 lines Allow filename completion on zero-length modules, - remove a memory leak, remove a file descriptor leak, and make - filename completion thread-safe. Patched and tested by tilghman. - (Closes issue #11681) ........ - -2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan - 2008) | 3 lines use the \retval doxygen command properly ........ - - * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan - 2008) | 5 lines Part 1 of N of adding doxygen comments to - app_queue. I picked some of the most common functions used (which - also happen to be some the biggest/ugliest functions too) to - document first. I'm pretty new to doxygen so criticism is - welcome. ........ - -2008-01-08 23:51 +0000 [r97305] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Add a new flag 'd' (with optional context) - permitting any extension within that context to be entered as a - new extension during the playback of a voicemail greeting. Patch - inspired by bluecrow76, by tilghman. (Closes issue #7063) - -2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo <rizzo@icir.org> - - * channels/console_board.c: add copyright (most of this code was - written by Marta Carbone), remove some unused code, add/clarify - some comments. - - * images/kpad2.jpg: Add the annotation for the textarea used for - messages, and also change the background from white to something - different to show that we can make use of fonts with transparent - background. - - * images/font.png (added): add a font suitable for use with the - console GUI. The background of this particular image is - transparent so we can preserve the original background when we - draw strings. - - * channels/console_gui.c, channels/console_video.c, - channels/console_board.c (added), channels/Makefile: add support - for textareas, used for various dialog windows on the gui. The - main code to implement the textarea is in console_board.c, and - uses a simple png image with the font, blitting characters on the - designated areas of the main screen. Additionally we provide some - annotations in the image used as a skin to indicate which areas - are used for text messages. (images will be committed - separately). At the moment the dialog area is only used to - display a running counter, just as a proof of concept. - -2008-01-08 21:56 +0000 [r97248] Terry Wilson <twilson@digium.com> - - * apps/app_queue.c: Initialize new variable to NULL - -2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the - option of specifying a second interface in a member definition - for a queue. app_queue will monitor this second device's state - for the member, even though it actually calls the first - interface. This ability has been added for statically defined - queue members, realtime queue members, and dynamic queue members - added through the CLI, dialplan, or manager. (closes issue - #11603, reported by acidv) - -2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson <oej@edvina.net> - - * channels/chan_console.c: Change reference to external library so - it appears on the extref listing - http://www.asterisk.org/doxygen/trunk/extref.html - - * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is - out with bug fixes. - -2008-01-08 20:56 +0000 [r97198] Tilghman Lesher <tlesher@digium.com> - - * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) - | 3 lines Increase constants to where we're less likely to hit - them while debugging. (Closes issue #11694) ........ - -2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: One line documentation ftw! - - * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 - lines Fix various DTMF issues in chan_mgcp. (closes issue #11443) - Reported by: eferro Patches: - dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license - 337) ........ - -2008-01-08 20:45 +0000 [r97193] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan - 2008) | 9 lines Making some changes designed to not allow for a - corrupted mailstream for a vm_state. 1. Add locking to the - vm_state retrieval functions so that no linked list corruption - occurs. 2. Make sure to always grab the persistent vm_state when - mailstream access is necessary. 3. Correct an incorrect return - value in the init_mailstream function. (closes issue #11304, - reported by dwhite) ........ - -2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Move common code for setting T38 - capabilities and fix a bug with fax detection in the SIP RTP read - callback. It's still sort of silly... but more on that later. - (closes issue #11239) Reported by: dimas Patches: - sipt38prop.patch uploaded by dimas (license 88) - - * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan - 2008) | 4 lines If no group has been provided to the GROUP_COUNT - dialplan function then use the first one specific to the channel. - (closes issue #11077) Reported by: m4him ........ - -2008-01-08 19:06 +0000 [r97125] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) - | 3 lines Apply multiple crash fixes, found in issue #11386, but - not completely closing that issue. ........ - -2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4 - lines Make app_queue calls work with directed pickup. (closes - issue #11700) Reported by: jbauer ........ - - * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically - appear in extconf. (closes issue #11703) Reported by: dmartin - -2008-01-07 23:03 +0000 [r96988] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c: add support for cropping the keypad image - while displaying it. This way it can contain additional elements - (e.g. fonts, buttons, widgets) without having to use a zillion - files to store them. - -2008-01-07 22:31 +0000 [r96987] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Explicitly make literal constants long - where they are expected to be. - -2008-01-07 21:12 +0000 [r96936] Jason Parker <jparker@digium.com> - - * main/config.c: Display a message if no config mappings are found - with "core show config mappings". Closes issue #11704, patch by - kshumard. - -2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Document some weird casting magic that's - necessary to interface with the c-client - - * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding - user-configurable TCP timeout settings to IMAP voicemail. This - could go a long way towards preventing unexplainable hangs - experienced by people. In the case of MWI hangs, this also will - mean that the SIP port isn't blocked anymore. (closes issue - #11665, reported by yehavi) - -2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant <russell@digium.com> - - * /, configs/extensions.conf.sample: Merged revisions 96932 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r96932 | russell | 2008-01-07 14:47:52 -0600 - (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 - Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com - ........ ................ - - * configs/http.conf.sample: Add a note about viewing the default - set of documentation using the built-in http server - - * Makefile: If the HTML documentation exists, install it in the - static-http/docs directory so that it can be viewed through the - Asterisk http server if it is turned on. - - * build_tools/prep_tarball: Build the HTML version of the doc files - for tarballs, as well - - * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) | - 3 lines Don't crash if something happens when setting up an SMDI - interface and it gets destroyed before the SMDI port handling - thread gets created. ........ - -2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming <kpfleming@digium.com> - - * formats/format_sln16.c (added): add a file-format driver for - 16KHz signed linear... which may or may not work - -2008-01-07 15:52 +0000 [r96858] Joshua Colp <jcolp@digium.com> - - * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck - manager commands from loader.c to manager.c. Previously they - would get registered twice because of the way manager.c operates. - (closes issue #11699) Reported by: caio1982 Patches: - manager_module_commands1.diff uploaded by caio1982 (license 22) - -2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c: update comments to reflect reality (or at - least planned behaviour). minor code cleanups - - * channels/console_gui.c: resolve a load-time problem avoiding a - call to console_do_answer. On passing, fix dialling from the - keypad. - -2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant <russell@digium.com> - - * res/snmp/agent.c: Convert this file over the new method of - getting the Asterisk version. (I don't have this building on this - machine, so caio1982 on IRC is going to test it for me. :) ) - - * Makefile, funcs/func_version.c, main/manager.c, - channels/chan_sip.c, main/Makefile, build_tools/make_version_c - (added), include/asterisk/version.h (added), res/res_agi.c, main, - main/http.c, build_tools/make_version_h (removed), - include/asterisk, main/asterisk.c: Now that the version.h file - was getting properly regenerated every time the svn revision - changed, every module that used the version was getting rebuilt - after every svn update. This severly annoyed me pretty quickly, - so I have improved the situation. Now, instead of generating - version.h, main/version.c is generated. version.c includes the - version information, as well as a couple of API calls for modules - to retrieve the version. So now, only version.c will get rebuilt, - and the main asterisk binary relinked, which is must faster than - rebuilding http.c, manager.c, asterisk.c, relinking the asterisk - binary, chan_sip.c, func_version.c, res_agi ... The only minor - change in behavior here is that the version information reported - by chan_sip, for example, is the version of the Asterisk core, - and not necessarily the Asterisk version that the chan_sip module - came from. - - * main/pbx.c: Print out the name of a function being registered in - color, just like the name of applications when they get - registered. - - * UPGRADE.txt: Add a note about changing modules.conf since another - console channel driver is now present that can not be used at the - same time as chan_alsa or chan_oss. - - * channels/chan_console.c: Add the URL to the home page for - portaudio. Also add the location of the svn repository to check - out portaudio v19. - - * /, main/devicestate.c: Merged revisions 96644 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | - 2 lines Don't pass an empty string as the device name. ........ - -2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_usbradio.c: improve chan_usbradio to use - indications just like chan_alsa/chan_oss do now - -2008-01-04 23:12 +0000 [r96576] Tilghman Lesher <tlesher@digium.com> - - * /, main/devicestate.c: Merged revisions 96575 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) - | 7 lines Fix the problem of notification of a device state - change to a device with a '-' in the name. Could probably do with - a better fix in trunk, but this bug has been open way too long - without a better solution. Reported by: stevedavies Patch by: - tilghman (Closes issue #9668) ........ - -2008-01-04 22:57 +0000 [r96574] Jason Parker <jparker@digium.com> - - * /, res/res_features.c: Merged revisions 96573 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600 - (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if - using PARKINGEXTEN and the slot is full. Issue 11237, patch by - me. ........ - -2008-01-04 19:35 +0000 [r96547] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) - | 4 lines If you change the bindaddr in sip.conf to a non-bound - address and reload, sip goes kablooie. Reported and patched by: - one47 (Closes issue #11535) ........ - -2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, - configure.ac, acinclude.m4: [commit message] (closes issue - #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff - uploaded by tzafrir (license 46) (modified by me and added - configure script support) - -2008-01-04 17:19 +0000 [r96499] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over - unsecured network connections only. This authentication mechanism - is implemented under the iksemel API, which makes use of GnuTLS, - whereas we use OpenSSL. Note : there's ongoing dicsussion at the - SASL IETF WG in order to deprecate SASL DIGEST-MD5, see - http://ietfreport.isoc.org/ids-wg-sasl.html. - -2008-01-04 16:21 +0000 [r96450] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | - 7 lines Make use of the temporary channel pointer while the pvt - is unlocked. (closes issue #11675) Reported by: flefoll Patches: - chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll - (license 244) ........ - -2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: we have to *always* use a completely silent 'make' - invocation for generating the module embedding rules - - * Makefile: there was no reason to add this define for non-Solaris - platforms - -2008-01-03 22:46 +0000 [r96395] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | - 3 lines Don't crash if the iax2 pvt structure has been destroyed - before we get to this point (closes issue #11672, reported by - snuffy, patched by me) ........ - -2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/channel.h: Document recent API addition - - * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03 - Jan 2008) | 4 lines Missed initialization caused crash. Reported - and fixed by: tiziano (Closes issue #11671) ........ - - * main/channel.c: Allow the uniqueid to be used for searching for a - channel in the list. Reported and initially patched by: - michael-fig (Closes issue #11340) - -2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, tests/Makefile (added), tests/test_skel.c (added), - tests (added): add some simple infrastructure for modules to be - used for testing parts of Asterisk - - * channels/answer.h (removed), channels/ring10.h (removed), - channels/busy.h (removed), channels/ringtone.h (removed), - channels/Makefile, channels/chan_oss.c, channels/gentone.c - (removed), channels: eliminiate sound_thread() and other stuff - from chan_oss since Asterisk indications can handle it remove - gentone and all the headers containing tones that are no longer - needed - - * channels/chan_alsa.c: coding guidelines cleanup remove background - thread and all sound generation mechanisms, as the built-in - indications can handle everything that is needed - -2008-01-03 14:47 +0000 [r96221] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 - Jan 2008) | 1 line when overlapdial was used and no number was - dialed, the call was dropped, now we just jump into the s - extension, which makes a lot more sense. ........ - -2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Add coordination between AMI and AGI applications, - with an asyncagi method Feature proposed and patched by: moy - (Closes issue #11282) - - * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility - fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669) - -2008-01-02 23:48 +0000 [r96103] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan - 2008) | 4 lines We need to reset the membername to NULL on each - iteration of this loop, otherwise the result is that multiple - members can have the same name, since the variable was not reset - on each iteration of the loop. ........ - -2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant <russell@digium.com> - - * channels/chan_console.c: Add support for generating a ringing - sound on an incoming call. This is a bit of a hack. It just asks - the core to generate the same tone that it would when you hear - ringback when making an outbound call. But hey, it works, and you - get the localized ring tone for the appropriate language set on - the channel. - - * channels/chan_console.c: Note that this module doesn't actually - play a ringing sound for an incoming call ... oops - - * channels/chan_console.c: Show the correct CLI command to answer - the call - -2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: actually parse and store echocan parameters - from zapata.conf... this *should* work <G> - -2008-01-02 22:40 +0000 [r96071] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't - use AST_C_DEFINE_CHECK for the two pthread things that may not - actually be definitions, they could be enums for example. - -2008-01-02 22:29 +0000 [r96028] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c: Add curly braces around a compound if - statement so that trunk will build properly - -2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: another - checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS - ioctl if it is present, but doesn't parse any supplied parameters - yet (this implementation is not very memory efficient as the - parameters and their values will be duplicated for each channel - that has the same settings, but we can worry about that later - once it is working) - -2008-01-02 21:49 +0000 [r96018] Russell Bryant <russell@digium.com> - - * main/libresample/include/libresample.h: Add doxygen documentation - to libresample.h while it's still fresh on my mind - -2008-01-02 21:08 +0000 [r95994] Mark Michelson <mmichelson@digium.com> - - * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c, - apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo, - bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue - #11668, reported and patched by mvanbaak) - -2008-01-02 20:26 +0000 [r95947] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 - lines Allocate a SIP refer structure when performing a transfer - using BYE with Also so that the transfer information is properly - stored. (AST-2008-001) (closes issue #11637) Reported by: - greyvoip ........ - -2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Since ',' is the standard argument separator in - trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of - AST_NONSTANDARD_APP_ARGS for determining member data. - - * include/asterisk/app.h: Fix a typo in a comment. - AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'. - -2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: clean up hwgain CLI command and improve docs - for swgain CLI command - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate - the macro being checked for, but just check for its existence - finish implementation of check for Zaptel HWGAIN support add - check for Zaptel ECHOCANCEL_PARAMS support - - * codecs/Makefile, include/asterisk/libresample.h (added), - codecs/codec_resample.c: and now just to keep the libresample - party going... if the functions from libresample are going to be - in the main Asterisk binary, it makes sense for the header that - defines them to be available without any special CFLAGS and to - out-of-tree modules building against /usr/include/asterisk - - * channels/chan_zap.c: umm... this did not compile on x86-64, and - could not possibly have worked on any platform as it was passing - string pointers to a function expecting ints - -2008-01-02 18:05 +0000 [r95891] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan - 2008) | 9 lines A change to improve the accuracy of queue logging - in the case where a member does not answer during the specified - timeout period. Prior to this change, there was a small chance - that the member name recorded in this case would be blank. Also - prior to this change, if using the ringall strategy, if no one - answered the call during the specified timeout, the member name - listed in the queue log would randomly be one of the members that - was rung. (closes issue #11498, reported and tested by hloubser, - patched by me) ........ - -2008-01-02 17:38 +0000 [r95888] Jason Parker <jparker@digium.com> - - * apps/app_osplookup.c: Update osplookup documentation to use - commas instead of pipes. Closes issue #11666, patch by Laureano. - -2008-01-02 16:20 +0000 [r95864] Russell Bryant <russell@digium.com> - - * main/Makefile, main/translate.c: For some odd reason, the last - set of libresample build changes from Kevin did not work for - everyone, but it did for some. This set of changes makes trunk - start again for those having problems. Instead of building - libresample as a static library, it just links the object files - in directly with the asterisk binary. - -2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: fix some long-time breakage that kept - chan_misdn from being embedded - - * channels/Makefile: use the proper technique for including - submodules so that embedding will work - - * CHANGES: note that chan_console requires portaudio v19 - - * configure, configure.ac: actually check for a function present in - libiconv (don't know how this test could have worked before) and - don't do the check on Linux/GNU systems because libiconv is not - present there and attempting to link with '-liconv' always fails - (it's not necessary as the iconv functionality is always - available) - - * main/libresample/src/filterkit.h, - main/libresample/src/resample.c, - main/libresample/win/libresample.dsp, main/libresample/configure, - main/libresample/Makefile.in, res/Makefile, - main/libresample/configure.in, main/libresample/src, - main/libresample/tests/testresample.c, - main/libresample/win/libresample.vcproj, - main/libresample/tests/compareresample.c, main/libresample/tests, - codecs/codec_resample.c, res/res_resample.c (removed), - main/libresample/README.txt, main/libresample/src/resamplesubs.c, - main/libresample/tests/resample-sndfile.c, - main/libresample/src/configtemplate.h, - main/libresample/install-sh, main/Makefile, main/translate.c, - main/libresample/include, main/libresample/src/resample_defs.h, - codecs/Makefile, main/libresample/config.guess, - main/libresample/config.sub, main/libresample/win, - main/libresample/LICENSE.txt, main/libresample (added), - main/libresample/Makefile.asterisk, build_tools/strip_nonapi, - res/libresample (removed), main/libresample/src/filterkit.c, - main/libresample/include/libresample.h: go back to including - libresample in the main Asterisk binary, but this time including - a small hack to ensure that it does get linked in (and also - modify the strip_nonapi script to leave the resample_<foo> - symbols alone) - -2008-01-02 11:34 +0000 [r95794] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Set stream flags to zero upon initialization. - When the XMPP over TLS/SSL connection resets for some reason, it - is wrongly believed as being secured, which makes the - re-connection process endlessly fail. This was reported by - mvanbaak in issue #11644. - -2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo <rizzo@icir.org> - - * main/loader.c: some cleanup of this code while I am trying to - debug a problem with gdb dying while debugging asterisk. The - problem seems to be related with a race in the handling of - module_list, which in turn is triggeded by calling dlopen() on a - system which uses initializers to create locks. - - * include/asterisk/module.h: There are three instances of the - module definition macros, which make maintaining this file very - error prone. This commit merges the embedded and !embedded - versions, and fixes the C++ version. Eventually we should move to - a single version of the macro. Too bad C++ doesn't like the - C-style struct initializers .foo = some_value - -2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant <russell@digium.com> - - * res/libresample/src/resample_defs.h, - res/libresample/src/resample.c: Don't make libresample print out - debugging output - - * main/translate.c: Make the translation table show slin16 - - * apps/app_meetme.c: fix a spacing issue introduced in revision - 95443. - - * main/Makefile, res/libresample/README.txt, res/Makefile, - res/libresample/install-sh, res/libresample/configure, - res/libresample/Makefile.in, res/libresample/include, - codecs/Makefile, res/libresample/configure.in, - res/libresample/src, res/libresample/config.guess, - main/libresample (removed), res/libresample/config.sub, - res/libresample/win, codecs/codec_resample.c, - res/libresample/LICENSE.txt, res/libresample (added), - res/libresample/Makefile.asterisk, res/libresample/tests, - res/res_resample.c (added): Instead of linking libresample into - the main Asterisk binary, build it as res_resample, and mark - codec_resample as dependent upon res_resample. This prevents the - linker from optimizing away libresample, and also makes it so the - libresample code isn't linked in to multiple places. (I have - another module in a branch that needs it, too.) - -2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c: call directly the cli command to - implement hangup. - - * channels/vcodecs.c: prevent a panic when destroying a channel - with no incoming video. - - * channels/console_video.c: remove a leftover sleep(1) used for - debugging - -2008-01-01 23:09 +0000 [r95648] Joshua Colp <jcolp@digium.com> - - * codecs/Makefile: Fix building of codec_resample on platforms - other then Cygwin. On everything else it actually gets built - after codec_resample, so you can't exactly link it in since it - doesn't exist. - -2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo <rizzo@icir.org> - - * codecs/Makefile, codecs/codec_resample.c: make codec_resample - build on __CYGWIN__, and make it load on FreeBSD (and probably - other systems as well). Both need libresample.a to be specified - in the linking phase, and cygwin needs <float.h> as other BSD. - The checks for OS-specific headers should really be moved to some - common header though. - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, - funcs/func_iconv.c, makeopts.in: implement "configure" checks for - libiconv, and add the iconv dependency for func_iconv. This fixes - some build issues on CYGWIN and FreeBSD and probably other - platforms where libiconv is not there by default - -2007-12-31 23:44 +0000 [r95578] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, /: Merged revisions 95577 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec - 2007) | 9 lines Avoiding a potentially bad locking situation. - ast_merge_contexts_and_delete writelocks the conlock, then calls - ast_hint_extension, which attempts to readlock the same lock. - Recursion with read-write locks is dangerous, so the inner lock - needs to be removed. I did this by copying the "guts" of - ast_hint_extension into ast_merge_contexts_and_delete (sans the - extra lock). (this change is inspired by the locking problems - seen in issue #11080, but I have no idea if this is the - problematic area experienced by the reporters of that issue) - ........ - -2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant <russell@digium.com> - - * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD. - Also, add some other platforms as they are likely the same. - - * channels/chan_console.c: Update chan_console to natively use a 16 - kHz sample rate. If it is talking to an 8 kHz endpoint, then - codec_resample will automatically be used to properly resample - the audio before sending it to/from chan_console. - - * main/libresample/src/filterkit.h, main/libresample/README.txt, - main/libresample/tests/resample-sndfile.c, - main/libresample/src/resamplesubs.c, main/Makefile, - main/libresample/install-sh, - main/libresample/src/configtemplate.h, - main/libresample/src/resample.c, - main/libresample/win/libresample.dsp, main/libresample/configure, - main/libresample/Makefile.in, main/libresample/include, CHANGES, - main/libresample/src/resample_defs.h, - main/libresample/configure.in, main/libresample/src, - main/libresample/config.guess, codecs/Makefile, - main/libresample/tests/testresample.c, codecs/slin_resample_ex.h - (added), main/libresample/config.sub, main/libresample/win, - main/libresample/win/libresample.vcproj, - main/libresample/LICENSE.txt, main/libresample (added), - main/libresample/Makefile.asterisk, main/libresample/tests, - main/libresample/tests/compareresample.c, codecs/codec_resample.c - (added), main/libresample/src/filterkit.c, - main/libresample/include/libresample.h: Merge changes from - team/russell/codec_resample This commit imports libresample for - use in Asterisk. It also adds a new codec module, codec_resample. - This module uses libresample to re-sample signed linear audio - between 8 kHz and 16 kHz. It also provides an alternative for - converting between 16 kHz G.722 and 8 kHz signed linear when - using G.722, which will likely be useful as some people have - complained about volume issues when the current codec_g722 - converts to 8 kHz signed linear. But, to test this, you will have - to disable the g722-to-slin and g722-to-slin16 translators in - codec_g722.c. - -2007-12-31 20:33 +0000 [r95490] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007) - | 3 lines Allow the default "0" to be returned if the STAT fails - (Closes issue #11659) ........ - -2007-12-31 18:46 +0000 [r95443] Mark Michelson <mmichelson@digium.com> - - * apps/app_meetme.c: Fix a compiler warning (closes issue #11658, - reported and patched by eliel) - -2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant <russell@digium.com> - - * configs/console.conf.sample (added), configs/modules.conf.sample, - channels/chan_console.c (added), CHANGES: Merge the main set of - changes from team/russell/chan_console. Add a new console channel - driver, chan_console, which is a console channel driver that uses - portaudio as a cross platform audio interface. It was written to - provide a console channel driver that works with Mac CoreAudio, - but it supports a number of other audio interfaces, as well, - including OSS and ALSA. It could one day be the single console - channel driver, but does not yet have as many features as - chan_oss. - - * include/asterisk/channel.h: fix a spelling error in a comment - - * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets - you set a config variable to a string field. (from - team/russell/chan_console) - - * configure, include/asterisk/autoconfig.h.in: Regenerate configure - script to include check for portaudio. - - * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add - configure script checking for portaudio. - -2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo <rizzo@icir.org> - - * channels/vcodecs.c, channels/console_video.c, channels/Makefile, - channels/console_video.h, channels/vgrabbers.c (added): Move - grabbers definitions to a separate file, vgrabbers.c, so it is - easier to add more entries. This required moving struct grab_desc - to the common header, and adding an entry in the Makefile. On - passing, cleanup some comments and file headers (some are still - missing). - - * channels/console_gui.c, channels/console_video.c: virtualize the - interface for video grabbers, which should make it easier to add - support for more grabbers (V4L2, firewire, and so on). - - * channels/console_video.c: Add a few entries up to 1408x1152 in - the table of known video resolutions. This makes it very - convenient to enlarge images using the right-click on the video - window. - - * channels/vcodecs.c, channels/console_video.c: change the - interface of video encapsulation routines, they only need the - buffer and mtu as input. - - * channels/console_gui.c, channels/vcodecs.c, - channels/console_video.c, channels/console_video.h: various - rearrangements and renaming of console_video stuff - -2007-12-28 18:39 +0000 [r95233] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: The diff for this change looks really bad, but - all I did here was decrease the indentation of most of the - queue_exec function by reversing the logic of an if statement. - This change makes the function comply better with the coding - guidelines. Since this change is purely a cosmetic change to the - code, I am only committing the change to trunk. - -2007-12-28 18:26 +0000 [r95192] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | - 6 lines Remove duplicate increment of the header count in the - add_header() function. (closes issue #11648) Reported by: makoto - Patch provided by sergee, committed patch by me, inspired by - comments from putnopvut ........ - -2007-12-28 16:12 +0000 [r95167] Mark Michelson <mmichelson@digium.com> - - * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel - name is printed in verbose messages maximumWordLength option - added. Duration of words that do not meet the minimum word - duration will be logged The duration of pre-greeting silence will - be logged Only consider us in the greeting if we actually - detected a valid word duration. (closes issue #11650, reported - and patched by davevg) - -2007-12-28 08:57 +0000 [r95139] Luigi Rizzo <rizzo@icir.org> - - * channels/console_video.c: fix a small bug in printing out - geometries - wrong input. - -2007-12-28 00:17 +0000 [r95096] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec - 2007) | 8 lines I found a bug while browsing the queue code and - managed to reproduce it in a small setup. If a queue uses the - ringall strategy, it was possible through unfortunate coincidence - for a single member at a given penalty level to make app_queue - think that all members at that penalty level were unavailable and - cause the members at the next penalty level to be rung. With this - patch, we will only move to the next penalty level if ALL the - members at a given penalty level are unreachable. ........ - -2007-12-27 23:32 +0000 [r95073] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove - more unnecessary casts for NULL. main/say.c is a big offender in - this respect. - -2007-12-27 23:28 +0000 [r95070] Jason Parker <jparker@digium.com> - - * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and - document it as well. Closes issue #11645, patch by Laureano. - -2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo <rizzo@icir.org> - - * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL - does not need to be cast to (char *) - - * channels/chan_oss.c: remove useless casts - -2007-12-27 21:41 +0000 [r95025] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 95024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | - 9 lines Don't report a syntax error when an empty string is - passed to ast_get_group. Just return 0. (closes issue #11540) - Reported by: tzafrir Patches: group_empty.diff uploaded by - tzafrir (license 46) -- slightly changed by me ........ - -2007-12-27 20:11 +0000 [r94978] Mark Michelson <mmichelson@digium.com> - - * /, main/io.c: Merged revisions 94977 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec - 2007) | 3 lines Fixing a typo in a comment. ........ - -2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 - lines Include types.h in chan_h323 as without it it can not be - compiled on some operating systems like FreeBSD to name one. - (closes issue #11585) Reported by: sobomax Patches: - chan_h323.c.diff uploaded by sobomax (license 359) ........ - - * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4 - lines Use ast_strlen_zero to see if our_contact is set or not on - the dialog. It is possible for it to be a pointer to NULL. - (closes issue #11557) Reported by: FuriousGeorge ........ - -2007-12-27 17:26 +0000 [r94904] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/console_video.c: more - localization of gui stuff - -2007-12-27 17:18 +0000 [r94903] Mark Michelson <mmichelson@digium.com> - - * doc/manager_1_1.txt: Adding documentation for new manager actions - and events in app_queue - -2007-12-27 16:51 +0000 [r94902] Luigi Rizzo <rizzo@icir.org> - - * CHANGES: clarify the type of video support in chan_oss - -2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant <russell@digium.com> - - * codecs/codec_g722.c: I went looking for where we downloaded the - g722 implementation and came across these two links. So, I'm - adding them so they are available for reference later. - - * /, main/translate.c, include/asterisk/translate.h: Merged - revisions 94828-94829 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | - 9 lines Change ast_translator_best_choice() to only pay attention - to audio formats. This fixes a problem where Asterisk claims that - a translation path can not be found for channels involving video. - (closes issue #11638) Reported by: cwhuang Tested by: cwhuang - Patch suggested by cwhuang, with some additional changes by me. - ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 - Dec 2007) | 2 lines Use the constant that I really meant to use - here ... ........ - -2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson <oej@edvina.net> - - * funcs/func_dialplan.c: This function checks more than just - contexts... - - * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: - - This application needs better documentation and a RESULT code in - the dialplan. - -2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c, /: Merged revisions 94824 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) - | 2 lines make this comment explain the situation in an even more - explicit fashion ........ - -2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c: more steps to decouple the gui from the - rest of the code. - - * channels/console_gui.c, channels/console_video.c, - channels/console_video.h: Enable building the code even if SDL is - not present (similarly, SDL is also detected at runtime). Now we - should be able to stream video even without a rendering device - (useful for remote monitoring). - - * channels/console_gui.c, channels/console_video.c: more - localizations around sdl_setup - - * channels/console_gui.c: use fread instead of mmap to read in the - comment area from the keypad. fread is simpler and more portable, - and there is no performance gain in using mmap. - - * images/kpad2.jpg: update the region description with an empty - line at the beginning. - -2007-12-26 22:38 +0000 [r94818] Tilghman Lesher <tlesher@digium.com> - - * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans - than 32. Also, rearrange compiler flags so the most often used - flags appear closer to the top. Reported by: tzafrir Patch by: - tzafrir,tilghman (Closes issue #11528) - -2007-12-26 22:29 +0000 [r94817] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/console_video.c: another bunch - of gui localizations - -2007-12-26 22:14 +0000 [r94814] Jason Parker <jparker@digium.com> - - * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean - up the description and usage text a bit. Closes issue #11564, - patch by pnlarsson (with some extra cleanup by me). - -2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/console_video.c: more - localization of sdl stuff - - * channels/console_gui.c, channels/console_video.c, - channels/console_video.h: move more gui stuff into console_gui.c - -2007-12-26 20:49 +0000 [r94809] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /: Merged revisions 94808 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) - | 6 lines Workaround for what is probably a glibc bug (but we'll - see this crop up again and again, if we don't add the - workaround). Reported by: rolek Patch by: tilghman (Closes issue - #11601, closes issue #11426) ........ - -2007-12-26 20:02 +0000 [r94806] Jason Parker <jparker@digium.com> - - * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, - apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c, - apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c, - pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c, - pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use - defined return values in load_module in more places. (closes - issue #11096) Patches: pbx_config.c.patch uploaded by moy - (license 222) pbx_dundi.c.patch uploaded by moy (license 222) - pbx_gtkconsole.c.patch uploaded by moy (license 222) - pbx_loopback.c.patch uploaded by moy (license 222) - pbx_realtime.c.patch uploaded by moy (license 222) - pbx_spool.c.patch uploaded by moy (license 222) - app_adsiprog.c.patch uploaded by moy (license 222) - app_alarmreceiver.c.patch uploaded by moy (license 222) - app_amd.c.patch uploaded by moy (license 222) - app_authenticate.c.patch uploaded by moy (license 222) - app_cdr.c.patch uploaded by moy (license 222) - app_zapateller.c.patch uploaded by moy (license 222) - app_zapbarge.c.patch uploaded by moy (license 222) - app_zapras.c.patch uploaded by moy (license 222) - app_zapscan.c.patch uploaded by moy (license 222) - -2007-12-26 20:01 +0000 [r94805] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c, channels/vcodecs.c, - channels/console_video.c, channels/console_video.h: more - preparation for untangling of the various console_video stuff - -2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: Merged revisions 94801 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | - 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already - set before starting autoservice, remember it and ensure that the - channel has the same setting when autoservice gets stopped. - (pointed out by d1mas, patched up by me) ........ - - * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan - function, DIALPLAN_EXISTS(), which allows you to check for the - existence of a dialplan target. (closes issue #11579) Reported - by: irroot Patches: func_dialplan2.c uploaded by irroot (license - 52) -- Additional changes by me. - - * main/autoservice.c, /: Merged revisions 94797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | - 4 lines When a channel is in autoservice, mark a flag on the - channel that says that we only care about the END of a digit. - That way, no magic digit emulation stuff will happen when all - we're doing is queueing up END frames. ........ - - * main/channel.c: Leave a note for a minor bug that was pointed out - by d1mas - -2007-12-26 18:05 +0000 [r94795] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: Convert raw bits for callprogress bitfield - to use constants, for greater code clarity Reported by: dimas - Patch by: dimas (Closes issue #11280) - -2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 94793 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) | - 3 lines Don't try to send a parked call back to itself. (closes - issue #11622, reported by djrodman, patched by me) ........ - - * Makefile, /: Merged revisions 94789 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) | - 5 lines List include/asterisk/version.h as a .PHONY target - because we want the commands listed for this target to be - executed regardless of whether the file exists or not. This fixes - having the version not up to date when running from svn. (closes - issue #11619, reported by plack, fixed by me) ........ - - * main/autoservice.c, /: Merged revisions 94790 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | - 5 lines Don't store DTMF BEGIN frames while a channel is in - autoservice. It's just going to make ast_read() do a lot of extra - work when the channel comes back out of autoservice. (closes - issue #11628, patched by me) ........ - - * channels/chan_iax2.c: Fix a bug in peer handling that caused - multiple instances of a peer to end up in the peers container - after a reload. Somehow, this bug doesn't exist in 1.4 ... - (closes issue #11626) (reported by pnlarsson, additional info - from mvanbaak, fixed by me) - - * utils: update svn:ignore for astcanary - -2007-12-26 15:58 +0000 [r94782] Mark Michelson <mmichelson@digium.com> - - * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding - support for storing the queue log entries in a realtime backend. - (closes issue #11625, reported and patched by sergee) Thank you - very much to sergee for adding this new feature! - -2007-12-26 10:14 +0000 [r94774] Luigi Rizzo <rizzo@icir.org> - - * channels/console_gui.c (added), channels/vcodecs.c (added), - channels/console_video.c: Split console_video.c so that video - codecs and gui functions are in separate files (still #include'd - because of tangling in the data structures, but this is going to - be cleaned up). The video grabbing functions still need to be - moved to a separate file. - -2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher <tlesher@digium.com> - - * apps/app_pickupchan.c (added): Add pickup by channel (Closes - issue #11161) - - * channels/chan_zap.c, configs/zapata.conf.sample: Change the - abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF - character. Also, fix the documentation to match the code. - - * res/res_agi.c: Add channel thread ID to the information passed to - AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162) - -2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 94767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) - | 5 lines Race: we need to wait to queue a NewChannel event until - after the channel is inserted into the channel list. The reason - is because some manager users immediately queue requests from the - channel when they see that event and are confused when Asterisk - reports no such channel. (Closes issue #11632) ........ - - * /: Merged revisions 94763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007) - | 5 lines Another bit of bad logic in realtime_peer Reported by: - dimas Patch by: dimas (Closes issue #11631) ........ - -2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo <rizzo@icir.org> - - * channels/console_video.c, channels/console_video.h: support - sdl_videodriver to send output to x11/aalib/console - - * channels/console_video.c: move reading info from the keypad to a - separate function. Remove an unused keypad field and some - debugging messages. Adjust formatting on config file parsing - - * channels/console_video.c: make sure the minimum surface depth is - 16bpp so we can create YUVoverlays. With this change we can do - setenv SDL_VIDEODRIVER aalib and output to an ascii window (which - is still in an X11 window). If you also do unsetenv DISPLAY then - the output goes into the main asterisk window, unfortunately it - interferes with the normal output so you don't see much. In any - case, i don't think we are very far away from having a working - xterm videophone! - - * channels/Makefile: avoid rebuilding dependent files if the - generated busy.h and ringtone.h do not change. Ths masks (but - does not solve) a but that i am seeing in doing a 'gmake install' - without donig a 'gmake all' first. - -2007-12-23 01:38 +0000 [r94662] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007) - | 2 lines Argh... I suppose third time's the charm. ........ - -2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo <rizzo@icir.org> - - * configs/oss.conf.sample, channels/console_video.c: Change the - name of config file entries for keypad regions from - 'keypad_entry' to 'region'. Fix the example file accordingly. - Also make some fixes in the code do reset entries on reload of - the keypad. The recently committed kpad2.jpg has the correct - names. - - * images/kpad2.jpg (added): add a sample keypad (with annotations) - for console video - - * channels/console_video.c, channels/Makefile, channels/chan_oss.c, - channels/console_video.h (added): Build console_video support by - linking in, as opposed to including, console_video.c This will - ease the task of splitting console_video.c into its components - (V4L and X11 grabbers, various video codecs and packetizers, - SDL), as well as ease future extensions (e.g. additional video - sources, codecs and rendering engines). For the time being - nothing changes for users: video support is off by default, and - requires -DHAVE_VIDEO_CONSOLE on the command line to be included - (if SDL and FFMPEG are available). - -2007-12-21 21:19 +0000 [r94593] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Something I've been itching to do for a - while now. A minor optimization in app_voicemail. Since the - dtable in base_encode always gets populated with the same values - every time and never changes, make it static and const and only - initialize it once. Also, there's no reason to define - BASEMAXINLINE twice, so remove the redundant #define. - -2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: We should only clear this value if we have - to - - * channels/chan_zap.c: Commit non TCP transport part of #11506. - Includes numerous additional parameters, as well as RLT support - for DMS type switches - -2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson <mmichelson@digium.com> - - * res/res_config_sqlite.c: Store dates using local time instead of - UTC (closes issue #11610, reported and patched by - rbraun_performatique) - - * apps/app_queue.c: Fix a memory leak when reloading queue rules. - - * CHANGES: The one documentation source I forgot to update after - the merge of the queue-penalty branch was the CHANGES file. No - longer! - - * apps/app_voicemail.c: Lots of coding guidelines cleanup. - - * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec - 2007) | 8 lines Better quota support for using IMAP storage - voicemail (closes issue #11415, reported by jaroth) (closes issue - #11152, reported by selsky) Patch provided by jaroth ........ - -2007-12-21 20:12 +0000 [r94541] Jason Parker <jparker@digium.com> - - * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c, - codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, - codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c: - codecs.conf really shouldn't be mandatory.. it never had been - before, so let's go back to being optional. A big "thank you" to - pnlarsson on IRC for allowing me access to his system to debug - this. Closes issue #11584. - -2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec - 2007) | 5 lines The mail_copy c-client function does not expect a - full imap mailbox string, just the name of the mailbox. (closes - issue #11419, reported and patched by jaroth, with additional - patchwork from me) ........ - - * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in - trunk - - * main/dial.c, /: Merged revisions 94468 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec - 2007) | 6 lines Since we are freeing list elements within a list - traversal, we need to use the safe traversal and remove the item - from the list before freeing it. (closes issue 11612, reported by - dtyoo) ........ - -2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec - 2007) | 3 lines Removing a debug message I accidentally just - committed ........ - - * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec - 2007) | 5 lines Fixing Portuguese syntax for saying dates and - times. Also some coding guidelines cleanup. (closes issue #11599, - reported and patched by caio1982, coding guidelines cleanup by - me) ........ - -2007-12-21 15:14 +0000 [r94419] Tilghman Lesher <tlesher@digium.com> - - * /, main/asterisk.c: Merged revisions 94418 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007) - | 2 lines Fix for restart-as-user problem reported via the -dev - list ........ - -2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Moved the update of the queue_ent's rule list - to just before we try to call queue members. This allows for the - change in penalty levels to be executed at the most logical time - frame. - - * configs/queues.conf.sample, doc/tex/channelvariables.tex, - apps/app_queue.c, configs/queuerules.conf.sample (added): Merging - the queue-penalty branch. In short, this allows one to - dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced - QUEUE_MIN_PENALTY during a call depending on the amount of time - passed. The purpose is to allow the call to open up to more (or - maybe just different) members without the caller's losing his - place in the queue. See configs/queuerules.conf.sample for an - example of how to set up queue rules and - configs/queues.conf.sample for how to associate a rule with a - queue. Along with the functional changes, new CLI and manager - commands exist to show the rules defined and there is an - additional CLI command to reload the queue rules. Future - enhancements that may be made: support for realtime queue rules - and support for dynamically adding a rule through the manager or - CLI. Also a manager command to reload the queue rules (I'll - probably write this myself very soon). - - * apps/app_voicemail.c: The changes to header inclusion in trunk - broke compilation of app_voicemail when using IMAP storage. The - reason is that c-client has its own definitions for LOG_WARNING - and LOG_DEBUG, so we need to be sure to include asterisk's - definitions last so that we use the proper values in - app_voicemail. (closes issue #11437, reported by blitzrage, patch - suggested by blitzrage) - -2007-12-20 22:39 +0000 [r94320] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample: Add a bit more to the description of - the "mwimonitor" option. - -2007-12-20 22:28 +0000 [r94319] Steve Murphy <murf@digium.com> - - * build_tools/make_buildopts_h: closes issue #11287; thanks to - snuffy for this fix, which will surely make all solaris owners - shout praises to his name. - -2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r94256 | russell | 2007-12-20 14:22:22 -0600 - (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 - Dec 2007) | 5 lines Fix another potential seg fault ... (closes - issue #11606) Reported by: dimas ........ ................ - - * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) | - 10 lines Fix a deadlock in d-channel handling in chan_zap. This - deadlock was introduced by the fix to ensure that channels are - properly locked when handling channel variables. There were - sections of this code where the channel pvt was locked before the - channel lock, when in fact it _must_ be the other way around. - (closes issue #11582) Reported by: bugi ........ - -2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_usbradio.c, include/asterisk/config.h, - channels/console_video.c, channels/chan_oss.c: add some macros to - simplify parsing the config file, see description in config.h . - They are a variant of the set of macros i used in chan_oss.c, - structured in a way to be more robust to the presence of spurious - ';' - basically, they define wrappers for 'do {' and '} while - (0)', plus some helper functions to deal with simple cases such - as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix - (CV_ as 'Config Variable') tries to be easy to remember and has - been chosen to not conflict with other existing macros in the - tree. For the time being, I have only updated the three source - files in the tree that used the old M_* macros. Hopefully, more - files will be converted. NOTE: I understand that inventing my own - dialect of C is generally wrong; however, the lack of adequate - support in the language encourages lazy programming practices - (such as ignoring errors, bounds, etc.) and this increases the - chance of vulnerability in the code, especially because we are - parsing user input here. Hopefully, these macros and the use of - ast_parse_arg (in config.h) should encourage the programmer to - write more robust code. - - * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c, - utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify - http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that - paths and filename are writable by asterisk.c without causing - segfaults. This involves defining the variables as const char *, - and having them point to as static, writable buffer defined in - asterisk.c On passing, fix some errors in using these variables - in some files in utils/ , and in res/snmp/agent.c which was - redefining a variable without using paths.h (not applicable to - 1.4) - -2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the - set_member_penalty code. 2. Fix an error when checking the CLI - command for setting a member's penalty. 3. Fix a logging error if - the incorrect parameter was the queue name or interface. (closes - issue #11544, reported and patched by Laureano) - - * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec - 2007) | 6 lines Sox versions 13.0.0 and newer do not have - "soxmix" and instead use sox -m. res_monitor needs to use this if - the user does not have soxmix. (closes issue #11589, reported by - amessina, patch inspired by amessina but with a flourish from me) - ........ - -2007-12-19 22:51 +0000 [r94085] Russell Bryant <russell@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 94077 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) | - 4 lines Check for the existence of the soxmix application on the - target platform and have the result available in autoconfig.h. - (part of issue #11589) ........ - -2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Add 'voicemail reload' command. Reported - by: eliel Patch by: eliel (Closes issue #11365) - - * apps/app_waituntil.c (added): Add contributed WaitUntil app. - Original code by pprindeville, updated for trunk by tilghman. - (Closes issue #11487) - -2007-12-19 19:29 +0000 [r94029] Russell Bryant <russell@digium.com> - - * include/asterisk/time.h: Add a couple of new time API calls - - ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported - by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license - 88) - -2007-12-19 17:58 +0000 [r94002] Luigi Rizzo <rizzo@icir.org> - - * channels/console_video.c: Add instructions on how to generate - your own font. - -2007-12-19 17:31 +0000 [r93956] Joshua Colp <jcolp@digium.com> - - * /: Merged revisions 93955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2 - lines Make the 1.4 builders happy, ensure var is NULL. ........ - -2007-12-19 17:13 +0000 [r93952] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007) - | 3 lines Avoid segfault in chan_iax when peer isn't defined - (Closes issue #11602) ........ - -2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c, include/asterisk/strings.h: Add a new API function, - written at least twice in app_voicemail.c and likely in other - places too. This is quite useful when placing mail/html stuff in - config files. /*! \brief Convert some C escape sequences - (\b\f\n\r\t) into the equivalent characters. \brief s The string - to be converted (will be modified). \return The converted string. - */ char *ast_unescape_c(char *s); - - * include/asterisk/config.h, main/config.c: add support for - PARSE_DOUBLE, and remove identifiers for types not supported - (INT16 and UINT16) - -2007-12-19 09:20 +0000 [r93899] Olle Johansson <oej@edvina.net> - - * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too - large... - -2007-12-19 08:57 +0000 [r93898] Luigi Rizzo <rizzo@icir.org> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK() - and use it to check for NETSMP and GTK (GTK is not used thoug). - AST_EXT_TOOL_CHECK() could be used for checking curl status as - well, perhaps with a small addition because we currently seem to - require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE - entry in makeopts.in We don't have yet any macros for using - pkg-config to check for a specific package (right now there is - only gtk2+ in the category). - -2007-12-19 08:57 +0000 [r93897] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the - ability to specify the To: header in an outbound INVITE by adding - an exclamation mark to the dial string. This patch also exists - for 1.4 in the fixtoheader-1.4 branch and has been in production - for quite some time. - -2007-12-19 08:12 +0000 [r93875] Luigi Rizzo <rizzo@icir.org> - - * res/snmp/agent.c: make netsmp build under AST_DEVMODE. - Description, included in the source, is below. I should note that - the PACKAGE_* macros that asterisk defines in autoconfig.h are - not used anywhere in the tree so they should just be removed. /* - * There is some collision collision between netsmp and asterisk - names, * causing build under AST_DEVMODE to fail. * * The - following PACKAGE_* macros are one place. * Also netsnmp has an - improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H - instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define - it to 0, however this will fail * when the proper check is - implemented. */ No - -2007-12-19 07:01 +0000 [r93854] Olle Johansson <oej@edvina.net> - - * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for - starting remote Asterisk by naming the actual runtime socket - instead of pointing to configuration file with -C Reported by: - sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax - (license 359) doc changes by committer (closes issue #11598) - -2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard <dhubbard@digium.com> - - * apps/app_osplookup.c: add missing header file - -2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Making the canary error message a little more - obvious. - - * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a - canary process, for high priority mode (asterisk -p) to ensure - that if Asterisk goes into a busy loop, the machine will be - recoverable. We'd still need to do a restart to put Asterisk back - into high priority mode, but at least a reboot won't be required. - (Closes issue #11559) - -2007-12-18 21:13 +0000 [r93741] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move some warnings away to debug since some - devices send a packet with a silly string as a NAT keepalive - packet. - -2007-12-18 18:39 +0000 [r93672] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions - 93668 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600 - (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 - Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) - ........ ................ - -2007-12-18 18:20 +0000 [r93666] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/paths.h: remove a leftover line with only a '#' - (wonder why the compiler does not complain!) and variables that - are only used in asterisk.c - -2007-12-18 17:05 +0000 [r93626] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 93625 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec - 2007) | 6 lines Rework deadlock avoidance used in ast_write, - since it meant that agent channels which were being monitored had - one audio file recorded and one empty audio file saved. (closes - issue #11529, reported by atis patched by me) ........ - -2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c, - utils/ael_main.c, utils/extconf.c, main/asterisk.c, - res/res_monitor.c, utils/conf2ael.c: make configuration variable - const so they are not accidentally modified. This requires - casting the strings in asterisk.c when writing to them, so we do - it through a macro to do it consistently. - - * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c, - apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c, - main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c, - cdr/cdr_csv.c: remove unnecessary (char *) casts for - ast_config_AST_* variables. There are some left in the .flex - files, left to the maintainer... - - * build_tools/make_defaults_h, main/asterisk.c: Rename the macros - in defaults.h - they are not meant to be globally visible. - Document the fact that DEFAULT_TMP_DIR cannot be overridden from - the default configuration (this needs to be fixed, as you could - have a totally different spooldir configured at runtime, and yet - DEFAULT_TMP_DIR keeps the compile-time default). Remove two - unused entries for sounds and images. - - * Makefile.moddir_rules: make the code match documentation - now - you can specify multiple words in MODULE_PREFIX. - - * CREDITS: Name the people responsible for some recent - contributions to the tree. - - * Makefile: Two small changes: + document the difference between - "A=foo make ..." and "make A=foo ..." and suggest using - COPTS/LDOPTS if you need to use the second form to pass compiler - and loader flags; + define only in one place the environment used - to build stuff in menuselect/ - -2007-12-18 07:56 +0000 [r93557] Olle Johansson <oej@edvina.net> - - * doc/CODING-GUIDELINES: A minor update, caused by a recent bug - report ;-) - -2007-12-18 07:22 +0000 [r93536] Luigi Rizzo <rizzo@icir.org> - - * doc/CODING-GUIDELINES: small documentation update (nothing - important). - -2007-12-18 02:57 +0000 [r93514] Joshua Colp <jcolp@digium.com> - - * channels/chan_unistim.c: You... will... build! I say so and - therefore you will. - -2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor - cleanups - -2007-12-17 23:10 +0000 [r93464] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_unistim.c: fix building under cygwin. At this point - WINARCH should go away. - -2007-12-17 22:54 +0000 [r93405] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_unistim.c: remove some unnecessary includes - -2007-12-17 22:50 +0000 [r93390] Jason Parker <jparker@digium.com> - - * /, main/translate.c: Merged revisions 93381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 - lines What was I thinking when I wrote this masterpiece? -1 + 1 = - 0.. who woulda thunk it?. ........ - -2007-12-17 22:38 +0000 [r93380] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: surprising as it may be, chan_oss compiles - correctly under cygwin as well, provided you look for soundcard.h - in the right place... - -2007-12-17 22:29 +0000 [r93378] Joshua Colp <jcolp@digium.com> - - * /, main/utils.c: Merged revisions 93377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 - lines Do not try to access information about a lock when printing - out a trylock attempt. It is possible for the lock that it - references to no longer be valid. This would have caused - segfaults or deadlocks. (issue #BE-263) (closes issue #11080) - Reported by: callguy (closes issue #11100) Reported by: callguy - ........ - -2007-12-17 21:14 +0000 [r93337] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17 - Dec 2007) | 6 lines Today is tomorrow's yesterday, and - yesterday's tomorrow is today, and tomorrow's tomorrow is the day - after tomorrow, so who cares if you recycle anyway? If this - confuses you, that's nothing compared to what this fixes. ;-) - ........ - -2007-12-17 21:12 +0000 [r93335] Olle Johansson <oej@edvina.net> - - * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c, - channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions - 93182 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 - lines Issue 11574: Add dependencies on res_monitor and - res_features. I wonder if Asterisk can run at all without - res_features. My guess is that there's propably a lot of more - modules and the core that depends on it. Reported by: caio1982 - (closes issue #11574) ........ - -2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Removing some leftover debug messages from a - while back. - - * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec - 2007) | 6 lines We need to create the directory for a voicemail - user even if they are using IMAP storage since greetings are - stored in the filesystem. (closes issue #11388, reported by - spditner, patch by me inspired by a patch by spditner) ........ - -2007-12-17 18:07 +0000 [r93252] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6 - lines If a call is received with a called number IE containing - nothing go to the 's' extension. (closes issue #9099) Reported - by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by - Corydon76 (license 14) ........ - -2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming <kpfleming@digium.com> - - * utils: all created files need to be listed in the ignore property - - * channels/chan_unistim.c, build_tools/menuselect-deps.in, - configure, configure.ac, channels/Makefile, channels/chan_oss.c: - make the configure script detect that it is running on a Windows - platform, and report that information so that menuselect can use - it (all information that is used to decide whether to build - modules or not must be fed to menuselect so the user knows what - will be built and why... don't make module build decisions in the - makefiles, please) - - * Makefile: make using PRINT_DIR a little easier - -2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix usage of rtptimeout. It can be used - without rtpkeepalive, and the value can not be accessed directly - in the SIP pvt structure. All RTP related timeouts have to be - retrieved using the ast_rtp_* function calls. (closes issue - #11562) Reported by: ibc - - * channels/chan_unistim.c: If no timezone is available use the - default message. (closes issue #11576) Reported by: junky - - * channels/chan_unistim.c: Make chan_unistim actually be able to - unload. When creating a thread that you want to pthread_join you - have to explicitly create it as joinable, and also if using - pthread_cancel you have to have a pthread_testcancel to see if it - has been called. - -2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming <kpfleming@digium.com> - - * codecs, /, build_tools/make_version, - include/asterisk/autoconfig.h.in, configure.ac, apps, - Makefile.moddir_rules, res/Makefile, pbx/Makefile, - build_tools/prep_moduledeps (removed), channels/Makefile, cdr, - formats, Makefile, codecs/Makefile, funcs, apps/Makefile, - configure, build_tools/embed_modules.xml, cdr/Makefile, - build_tools/prep_tarball, makeopts.in, formats/Makefile, res, - pbx, channels, funcs/Makefile: Merged revisions 93180 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) - | 23 lines In - http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, - rizzo brought up some issues related to the way that the metadata - required for menuselect and the rest of the build system is - extracted from the source files. Since I had a few hours to kill - on an airplane today, I decided to improve this situation... so - now the system caches the extracted metadata and uses it to build - the menuselect 'tree' as much as it can. The result of this is - that when a single source file is changed, only the metadata for - that file needs to be extracted again, and the rest is used from - the cache files. I also reduced the number of forked processes - required to do the metadata extraction; it was actually possible - to do most of what we needed in the Makefiles themselves without - using any shell scripts at all! On my laptop, these changes - resulted in an 80% decrease in the time required for the - 'menuselect.makeopts' automatic check to occur after editing a - single source file. While doing this work I also cleaned up a few - minor things in the Makefiles, adding a check for 'awk' to the - configure script and changed all remaining places we use 'grep' - or 'awk' to use the ones found by the configure script, and - changed the 'prep_tarball' script to build the menuselect - metadata so that tarballs of Asterisk will include it and won't - require the user to wait while it is extracted after unpacking. - ........ - -2007-12-16 19:06 +0000 [r93173] Luigi Rizzo <rizzo@icir.org> - - * Makefile: menuselect.makeopts is not a .PHONY target - -2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson <oej@edvina.net> - - * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks, - dimas! (closes issue #11572) Reported by: dimas Patches: - dundilog-trunk.patch uploaded by dimas (license 88) - - * main/manager.c, CHANGES: Adding a new CLI command for "manager - reload", which is important now that you need to reload after - changes. Thanks YS. Reported by: ys Patches: - trunk93163_manager_reload.c.diff uploaded by ys (license 281) - (related to issue #11414) - - * main/manager.c, CHANGES: Change manager so that registered - accounts are stored in memory. This opens for a manager realtime - implementation. If you change accounts in manager.conf, you now - need to reload to activate the changes (deletions, additions). - This was not the case with 1.4. Reported by: ys Patches: - trunk93163_manager_reload.c.diff uploaded by ys (license 281) - (closes issue #11414) - - * CHANGES: Adding console_video to CHANGES. It's important that we - keep this file up to date, even with experimental stuff. - - * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample, - channels/chan_sip.c, include/asterisk/rtp.h, - include/asterisk/netsock.h, channels/iax2-provision.c, - UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample, - CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c, - configs/h323.conf.sample, configs/iax.conf.sample, - channels/chan_skinny.c, configs/mgcp.conf.sample, - configs/unistim.conf.sample, channels/chan_h323.c, - configs/iaxprov.conf.sample, pbx/pbx_dundi.c, - configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements - to QoS/CoS handling by IgorG - Refer to the proper documentation - - Implement separate signalling/media QoS/CoS in many channels - using RTP - Improve warnings and verbose messages - Deprecate - some old settings Minor modifications by me, a big effort from - IgorG. Thanks! Reported by: IgorG Patches: - qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) - Tested by: IgorG (closes issue #11145) - -2007-12-16 10:34 +0000 [r93162] Luigi Rizzo <rizzo@icir.org> - - * Makefile: use a simpler idiom for 'cmp -s ...' - -2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson <oej@edvina.net> - - * main/asterisk.c: Don't drop the first character randomly in long - listings in the CLI. Reported by: slavon Patches: - asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich - (license 176) Tested by: eliel (closes issue #9325) - - * configs/sip.conf.sample, CHANGES: Update documentation - - * channels/chan_sip.c, configs/sip.conf.sample: Make more timers - settable in SIP so that we can force timeout earlier on - non-responsive SIP servers. Thanks, jcmoore, for the patch! - Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt - uploaded by jcmoore (license 9) (closes issue #9771) - - * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo - after all. Didn't a clever guy once say "Compile before you - commit" ? :-) - -2007-12-15 08:10 +0000 [r93151] Russell Bryant <russell@digium.com> - - * include/asterisk/file.h: fix a typo from revision 93138 - -2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo <rizzo@icir.org> - - * configs/oss.conf.sample: configuration options related to video - support. - - * channels/console_video.c (added): Bring in video console support - for chan_oss (and later chan_alsa too). This is disabled in the - default build, you need to explicitly enable it compiling with - make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do - a video call with chan_oss, using the webcam (or X11 grabbing) as - local source, and rendering the incoming stream on your screen. - Currently supported formats are h261, h263, h263+, h264, mpeg4 - (all through the avcodec lib, part of ffmpeg). Incoming video is - on the left, outgoing video is on the right, while the center - displays a keypad (if configured so). Right clicking on the video - windows increases the size, center clicking reduces the size. - Dragging the mouse (with the left key) on the right window while - the X11 grabber is active moves the grab area. This is the result - of work by Sergio Fadda, Marta Carbone and myself, all properly - disclaimed to digium. Note, there is a lot of work left to do in - this module, including adding support for Video4LinuxV2 (I have - patches from Matteo Brancaleoni which should be integrated), and - making the GUI a lot more friendly than it is now (e.g. - supporting merging or switching among multiple sources, a text - window, and more). - - * channels/chan_oss.c: remove some redundant headers - - * include/asterisk/file.h: include mmap header if detected by - configure - -2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Resolve a compiler warning - - * apps/app_voicemail.c: Change places where the name "INBOX" was - hardcoded to use the imapfolder setting from voicemail.conf - instead. This commit will help to get issue #11415 moving towards - commitment. - -2007-12-14 21:09 +0000 [r93090] Tilghman Lesher <tlesher@digium.com> - - * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h: - Solaris compat fixes Reported by: snuffy Patch by: - snuffy,tilghman (Closes issue #11315) - -2007-12-14 19:31 +0000 [r93067] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: make something static - -2007-12-14 19:27 +0000 [r93066] Tilghman Lesher <tlesher@digium.com> - - * apps/app_privacy.c, UPGRADE.txt, CHANGES, - configs/privacy.conf.sample (removed): Remove use of privacy.conf - by the Privacy app. Reported by: eliel Patch by: eliel (Closes - issue #11344) - -2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to - increment the numbers used on the VERBOSITY_ATLEAST calls by 1. - Thanks to kpfleming for pointing this out. - - * include/asterisk/logger.h, main/pbx.c, main/manager.c, - funcs/func_timeout.c: Changed VERBOSITY_LEVEL to - VERBOSITY_ATLEAST to be more accurate. - - * include/asterisk/logger.h, main/pbx.c, main/manager.c, - funcs/func_timeout.c, main/logger.c: After reading Russell's - e-mail to the dev list stating that checking option_verbose is - not equivalent to the check done by ast_verb, I wrote a macro, - VERBOSITY_LEVEL, which does this check. I did a quick look in the - source and used this macro in some places where option_verbose - was used. I also converted some verbose messages in logger.c to - use ast_verb instead of ast_verbose. - -2007-12-14 18:24 +0000 [r93041] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: gcc 4.1.3 wants a union used here. - -2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant <russell@digium.com> - - * main/config.c: Print an error message if a #included file does - not exist - -2007-12-14 17:29 +0000 [r92999] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Publish the AGI events to manager. Reported by: - moy Patch by: moy,tilghman (Closes issue #11337) - -2007-12-14 15:59 +0000 [r92976] Mark Michelson <mmichelson@digium.com> - - * funcs/func_timeout.c: Reintroduce an optimization that was lost - when converting trunk to use ast_verb. - -2007-12-14 15:49 +0000 [r92939] Tilghman Lesher <tlesher@digium.com> - - * main/editline/sys.h: If malloc.h is included in a Solaris build, - the compilation breaks. Reported by: snuffy Patch by: snuffy - (Closes issue #11313) - -2007-12-14 15:18 +0000 [r92938] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 - lines Up the length of the format on the SIP channel since it can - now be rather long. (closes issue #11552) Reported by: - francesco_r ........ - -2007-12-14 15:14 +0000 [r92936] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 92933 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007) - | 5 lines Change help documentation to match actual behavior - (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman - (Closes issue #11548) ........ - -2007-12-14 15:08 +0000 [r92935] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 - Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state - setting and overlap_task thread starting. ........ - -2007-12-14 14:48 +0000 [r92913] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c, - res/res_features.c, apps/app_minivm.c, apps/app_amd.c, - res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c, - main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert - ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy - (Closes issue #11547) - -2007-12-14 01:25 +0000 [r92876] Mark Michelson <mmichelson@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13 - Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't - spam the CLI with messages about possible deadlocks. Instead just - print the intended single message every five seconds. (closes - issue 11537, reported and patched by dimas) ........ - -2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: When working with dates, use numeric form - whenever possible, as it's faster. Also, a bunch of coding - guidelines fixes. - - * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) - | 5 lines Properly initialize polarity statuses, so that they are - detected properly. Reported by: julianjm Patch by: julianjm - (Closes issue #10238) ........ - -2007-12-13 20:23 +0000 [r92811] Joshua Colp <jcolp@digium.com> - - * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c, - apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the - new ast_module_user_* functions in a few documentation places. - (closes issue #11533) Reported by: IgorG Patches: - oldmacroclean.v1.diff uploaded by IgorG (license 20) - -2007-12-13 20:14 +0000 [r92810] Jason Parker <jparker@digium.com> - - * main/pbx.c, /: Merged revisions 92809 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1 - line Make application help text a little more clear about the use - of extensions in a filename. ........ - -2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec - 2007) | 3 lines Prevent another potential fd leak ........ - - * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec - 2007) | 3 lines Prevent a possible fd leak. ........ - -2007-12-13 17:46 +0000 [r92779] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape - character, unless it really is an escape character. - -2007-12-13 16:23 +0000 [r92758] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Remove remnants of a poorly merged commit. - (92697) - -2007-12-13 15:40 +0000 [r92737] Doug Bailey <dbailey@digium.com> - - * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to - local time zone - -2007-12-13 00:18 +0000 [r92697] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c: - Merged revisions 92696 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600 - (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file, - we previous continued on with what was already loaded. We do not - want to do this (see bug below for details). This makes it so - that if a [ is found without a ], the entire config will fail, - and nothing in it will be loaded. Issue 10690. ........ - -2007-12-12 23:44 +0000 [r92676] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Revert an "optimization" that I added in - revision 89887, as the user who reported issue #11449 has - demonstrated that it actually was a performance hit on his - machine. I think that it is possible that it could still be a - benefit on systems under higher load, especially SMP systems, but - I don't have enough time or interest to find out at the moment. - (closes issue #11449) - -2007-12-12 21:22 +0000 [r92618] Jason Parker <jparker@digium.com> - - * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600 - (Wed, 12 Dec 2007) | 4 lines Don't increment user count until - after name has been recorded (if enabled). Issue 11048, tested by - pep. ........ - -2007-12-12 20:05 +0000 [r92594] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, main/logger.c, main/utils.c, - apps/app_mixmonitor.c: Conversions of free to ast_free, where - applicable, and several other formatting fixes. Reported by: - eliel Patch by: eliel,tilghman (Closes issue #11209) - -2007-12-12 19:50 +0000 [r92562] Russell Bryant <russell@digium.com> - - * res/res_features.c: Merged revisions 92556 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) | - 1 line resolve compiler warning ........ - -2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson <mmichelson@digium.com> - - * res/res_features.c: Same change to trunk as revision 92510. I'm - not sure why I merged this way, but I did. - -2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Correctly handle possible memory allocation - failure Reported by: eliel Patch by: eliel (Closes issue #11512) - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 92463 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007) - | 4 lines Test directly for the API that fixed AST-2007-026, to - ensure that older versions of PostgreSQL are no longer - acceptable. (Closes issue #11526) ........ - -2007-12-12 16:11 +0000 [r92444] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec - 2007) | 3 lines Removing an unused variable. ........ - -2007-12-11 22:20 +0000 [r92423] Olle Johansson <oej@edvina.net> - - * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c, - channels/ringtone.h, include/asterisk/ulaw.h, - include/jitterbuf.h, include/asterisk/manager.h, - include/asterisk/transcap.h, channels/misdn/isdn_lib.c, - channels/gentone.c, include/asterisk/zapata.h, - channels/misdn/isdn_lib.h, include/asterisk/doxyref.h, - channels/DialTone.h, channels/misdn/ie.c, - channels/misdn/chan_misdn_config.h, channels/iax2.h, - channels/misdn/portinfo.c, include/asterisk/udptl.h, - main/cygload.c, include/asterisk/translate.h: Doxygen updates, - formatting. misdn stuff needs a lot of doxygenification (Hello, - Qwell :-) ) - -2007-12-11 22:10 +0000 [r92422] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, - configure.ac: Trunk build would fail due to the nonexistence of - zaptel hwgain structures missing. Patched configure to check for - this stuff and put a #ifdef around the offending code in - chan_zap. Thanks to file for overseeing this. - -2007-12-11 21:58 +0000 [r92421] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: We need to set the address we want to match - against before we actually do the match.. Closes issue #11518. - -2007-12-11 21:46 +0000 [r92402] Mark Michelson <mmichelson@digium.com> - - * res/res_musiconhold.c: Removing a pointless memset. The memory - was just calloc'd, so the memory is already zeroed out - -2007-12-11 21:17 +0000 [r92401] Jason Parker <jparker@digium.com> - - * apps/app_controlplayback.c: Add variable to show which key was - pressed to stop playback. Issue #11377, initial patch by johan. - -2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp <jcolp@digium.com> - - * res/res_monitor.c: Only look to see if options are set if some - have been provided. (closes issue #11505) Reported by: Mike - Anikienko - - * main/global_datastores.c, /: Merged revisions 92363 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec - 2007) | 6 lines Fix potential memory leak with the dialed - interfaces list if another memory allocation fails. (closes issue - #11507) Reported by: eliel Patches: global_datastores.c.patch - uploaded by eliel (license 64) ........ - -2007-12-11 17:44 +0000 [r92324] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec - 2007) | 10 lines Fixing autofill to be more accurate. - Specifically, if calls ahead of the current caller were ringing - members (but not yet bridged) there could be available members - and waiting callers who would not get matched up. The member - availability checker was correctly determining the number of - available members in this scenario, but the queue itself did not - parallelly reflect this status on the pending calls. This commit - corrects the issue. (closes issue #11459, reported by - equissoftware, patched by me) ........ - -2007-12-11 16:29 +0000 [r92305] Russell Bryant <russell@digium.com> - - * include/asterisk/unaligned.h, main/event.c: * In unaligned.h, - remove some unnecessary casts and mark the arg of the - get_unaligned functions as const * In event.c, use - get_unaligned_uint32() in a couple of places to fix issues on - architectures that don't allow unaligned access - -2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson <oej@edvina.net> - - * include/asterisk/devicestate.h, include/asterisk/agi.h, - include/asterisk/astobj2.h, include/asterisk/extconf.h, - include/asterisk/io.h, include/asterisk/cdr.h, - include/asterisk/aes.h, include/asterisk/_private.h, - include/asterisk/localtime.h, include/asterisk/hashtab.h, - include/asterisk/callerid.h, include/asterisk/logger.h, - include/asterisk/doxyref.h, include/asterisk/app.h, - include/asterisk/adsi.h, include/asterisk/event.h, - include/asterisk/causes.h, include/asterisk/alaw.h, - include/asterisk/ast_expr.h, include/asterisk/dsp.h, - include/asterisk/mod_format.h, include/asterisk/ael_structs.h, - include/asterisk/astdb.h: A lot of doxygen updates - - * include/asterisk/frame.h: Doxygen updates - -2007-12-10 20:18 +0000 [r92243] Doug Bailey <dbailey@digium.com> - - * channels/chan_zap.c: Add CLI commands to dynamically set hw and - sw gains - -2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp <jcolp@digium.com> - - * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to - check_expr for platforms that need it. (closes issue #11484) - Reported by: snuffy - - * /, main/rtp.c: Merged revisions 92204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 - lines Add G729A as another possible payload name for G729. Some - devices use this instead of G729, which is perfectly normal since - the payload number itself is defined and can't be used by - anything else so the name doesn't matter that much. (closes issue - #11483) Reported by: revolution Patches: rtp.diff uploaded by - revolution (license 346) ........ - -2007-12-10 16:30 +0000 [r92203] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec - 2007) | 7 lines If there are no members in a queue, then the loop - where the datastore for detecting duplicate dialed numbers will - be skipped, meaning the datastore isn't created. This means that - when we try to free it, there's a crash. This stops that crash - from occurring. (closes issue #11499, reported by slavon, patched - by eliel) ........ - -2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1, - otherwise all PIDs greater than -1 will get the SIGHUP... and - that is bad. (closes issue #11453) Reported by: alanmcmillan - -2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Removing some LOG_DEBUG items - - * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 - lines Avoid reinvite race situations with two Asterisks trying to - reinvite each other in 1.4 and trunk. This patch implements - support for the 491 error code that Asterisk 1.4 generates on - situations where we get an incoming INVITE and already has one in - progress. Thanks to mavetju for reporting and to Raj Jain for an - excellent explanation of the problem. Patch by myself. Tested - with 8 Asterisk servers connected to each other in a training - network. Closes issue #10481 ........ - - * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra - headers in the voicemail users listing in manager 1.1. Update - documentation too. (closes issue #11495) Reported by: caio1982 - Patches: extra_vm_manager_info1.diff uploaded by caio1982 - (license 22) - -2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo <rizzo@icir.org> - - * build_tools/make_version, build_tools/make_version_h: - simplify/cleanup the scripts - - * utils/Makefile: remove relative paths and use ASTTOPDIR instead. - Give a default value to ASTTOPDIR if unset so we can at least do - a 'make clean' without too much trouble. The proper fix, however, - is to partition the top level Makefile in a 'setup' and a 'main' - part, in a way that the 'setup' part can be included from - subdirs' Makefiles and allow targets to be built without going - through the top level Makefile. - - * utils/clicompat.c: simplify this file - - * doc/CODING-GUIDELINES: add a bit of info on the build - infrastructure - - * Makefile: Fix the detection of modules installed from this build. - You can now add the path of local module subdirs from the command - line with make LOCAL_MOD_SUBDIRS= .... - - * codecs/Makefile, apps/Makefile, Makefile.moddir_rules, - cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, - formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules - the common instructions used to generate loadable and embedded - module lists. Individual Makefiles now are a lot simpler, - possibly as simple as this: -include - $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps - MODULE_PREFIX=cdr_ all: _all include - $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because - in a single directory we can combine various types of modules - (app_, cdr_, func_, ... ) by simply listing them in the - MODULE_PREFIX variable. The individual Makefiles can also create - list of modules to be excluded by listing them in the variablel - MODULE_EXCLUDE (see an example in channels/Makefile). With this - change it becomes trivial to integrate a directory with locally - created/modified sources into the main build. - - * Makefile, Makefile.moddir_rules: make the install target a bit - less noisy - - * Makefile: document usage of several exported variables - - * utils/Makefile: add hashtab.c to the list of files deleted - - * Makefile.moddir_rules: another place where ../ should have been - ASTTOPDIR - - * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile, - pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, - funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR - and not .. to reference the top level directory. - - * Makefile: Implement the outcome of a discussion on the -dev list - re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir - Cohen and Simon Perreault for extremely useful feedback: 1. - comment out the [directories] section the created asterisk.conf ; - you can set the correct defaults at build time using - INSTALL_PATH, so the repetition here is redundant and often - wrong. (The next step now is move asterisk.conf outside the - Makefile to asterisk.conf.sample, because there is little if - anything here that needs to be constructed at build/install - time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to - specify a path once if the two coincide. This should have no ill - side effects, because if you don't specify DESTDIR, you really - need INSTALL_PATH="" to set the correct defaults, and if you - specify DESTDIR the value is not overridden. The second part - required moving the 'export DESTDIR' right after the assignment - to prevent DESTDIR getting set by the export (this is documented - in the Makefile).o hopefully avoid the mistake)$ With this change - you can now do something like this from your source tree: make - INSTALL_PATH=/some/place install samples and then main/asterisk - -vdc which will pick up the correct config files and libraries - from /some/place - i.e. great for developers! - - * main/config.c: remove unused code, and simplify the logic for - #include/#exec (still a lot of cleanup needed here). - - * main/config.c: Implement comment_buffer and lline_buffer in terms - of the ast_str_*() API. I don't know if comment_buffers etc are - actually used at all... - - * main/config.c: unify some common code - - * main/config.c: normalize formatting - - * main/config.c: document a nice technique to exit from a block in - case of errors. - - * main/config.c: a little bit of documentation on how lines are - parsed. - - * utils/ael_main.c: normalize header order, and add a comment on - the need to clean up this file. - - * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need - netinet/in.h to be included before arpa/inet.h - -2007-12-07 23:32 +0000 [r91832-91891] Jason Parker <jparker@digium.com> - - * /, main/dsp.c: Merged revisions 91890 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600 - (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the - input frame if we return a different frame in ast_dsp_process. - Issue 11273, pointed out by dimas, with a patch by eliel. - ........ - - * pbx/pbx_lua.c, configs/extensions.lua.sample: Update - documentation for pbx_lua. Closes issue #11492, patch by - mnicholson. - -2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant <russell@digium.com> - - * /, main/utils.c: Merged revisions 91830 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) | - 5 lines Make the lock protecting each thread's list of locks it - currently holds recursive. I think that this will fix the - situation where some people have said that "core show locks" - locks up the CLI. (related to issue #11080) ........ - - * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07 - Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code. - The ast_mutex_init() function had the mutex attribute object - marked as static. This means that multiple threads initializing - locks at the same time could step on each other and end up with - improperly initialized locks. (found when tracking down locking - issues related to issue #11080) ........ - - * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07 - Dec 2007) | 6 lines I love fixing lock related errors in the lock - debugging code. That's about as ironic as it gets in Asterisk - programming land. Anyway, I spotted this bug while trying to - track down why systems are locking up and acting weird in issue - #11080. The mutex attribute object was marked as static in this - function when it should not have been. ........ - - * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | - 6 lines * Add channel locking around datastore operations that - expect the channel to be locked. * Document why we don't record - Local channels in the dialed interfaces list. * Remove the dialed - variable as it isn't needed. * Restructure some code for clarity - and coding guidelines stuff ........ - -2007-12-07 16:37 +0000 [r91782] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Fix a small typo in a comment. Closes issue - #11490 - -2007-12-07 16:28 +0000 [r91781] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | - 7 lines * Add channel locking around datastore operations that - expect the channel to be locked. * Document why we don't record - Local channels in the dialed interfaces list. * Handle memory - allocation failure. * Remove the dialed variable, as it wasn't - actually needed. * Tweak some formatting to conform to coding - guidelines. ........ - -2007-12-07 16:11 +0000 [r91779] Jason Parker <jparker@digium.com> - - * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c, - include/asterisk/pbx.h, main/cli.c: Add count of total number of - calls processed by asterisk during it's lifetime. Add number of - total calls and current calls to SNMP. Closes issue #10057, patch - by jcmoore. - -2007-12-07 16:11 +0000 [r91778] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: Merged revisions 91777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) | - 6 lines * Add a bit more of a verbose comment as to why a hangup - frame needs to be queued up if autoservice gets a NULL return - from ast_read(). * Make the process of queueing the hangup frame - more efficient by putting the frame where it is going to end up - and avoiding some locking and extra memory allocations and - freeing. ........ - -2007-12-07 15:40 +0000 [r91738] Mark Michelson <mmichelson@digium.com> - - * main/autoservice.c, /: Merged revisions 91737 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec - 2007) | 7 lines Hangups that happen during autoservice were not - processed appropriately. This is because a hangup actually causes - a NULL frame to be received, not a hangup frame. Queueing a - hangup if we receive a NULL frame during autoservice corrects - this problem (closes issue #11467, reported by jmls, patched by - me) ........ - -2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | - 2 lines Don't unlock the dialed_interfaces list until we're done - messing with the iterator. ........ - - * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | - 4 lines Allow dialing local channels from Queue() and Dial() - again. There was a slight flaw in the code to prevent call - forwards from looping that caused this problem. (related to issue - #11486) ........ - - * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | - 7 lines Fix in an issue in the call forwarding handling code that - was causing crashes on every call into a queue. I'm not entirely - sure about the logic in this part of the code, so I want to look - at it some more tomorrow. However, this makes it safe and keeps - it from crashing. (closes issue #11486, reported by adamg, - patched by me) ........ - -2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher <tlesher@digium.com> - - * /, main/rtp.c: Merged revisions 91637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) - | 5 lines At the end of a call, when we're reporting, RTCP may - already be partially torn down, so check for NULL dereference - Reported by: blitzrage Patch by: tilghman (Closes issue #11450) - ........ - - * channels/chan_zap.c: Add a manager event for PRI events: this - will help manager users detect when a D-channel goes down - - * main/cdr.c: If duration or billsec are not yet calculated, - calculate them on demand. - -2007-12-06 21:57 +0000 [r91598] Jason Parker <jparker@digium.com> - - * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3 - cdr module.. Closes issue #11070, patch by seanbright. - -2007-12-06 21:03 +0000 [r91579] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Handle allocation failure of the heard and - deleted arrays of the vm_state. (closes issue #11408, reported - and patched by jaroth) - -2007-12-06 20:52 +0000 [r91561] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007) - | 3 lines Properly escape cdr->src and cdr->dst and ensure we use - thread-safe escaping (Fixes AST-2007-026) ........ r90736 | - tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines - If both dbhost and dbsock were not set, a NULL deref could result - Reported by: xrg Patch by: tilghman (Closes issue #11387) - ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 - Dec 2007) | 5 lines Solaris requires the inclusion of - sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: - snuffy,tilghman (Closes issue #11430) ........ - -2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we clear these flags when libpri - is not installed - -2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp <jcolp@digium.com> - - * main/udptl.c, /: Merged revisions 91450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6 - lines Fix various in the udptl implementation. It could return - empty modem frames, have an incorrect sequence number on packets, - and display the wrong sequence number in the debug messages. - (closes issue #11228) Reported by: Cache Patches: udptl-4.patch - uploaded by dimas (license 88) ........ - - * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 - lines Add support for accepting and sending T.38 in the initial - INVITE. (closes issue #9402) Reported by: thdei ........ - -2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson <oej@edvina.net> - - * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation - for the massive manager changes to manager version 1.1 - - hopefully a more consistent manager interface. - - * main/manager.c: - The Ping Action - Now use Response: success - - New header "Ping: pong" :-) - The Events action - Now use - Response: Success - The new status is reported as "Events: On" or - "Events: Off" - Report if manager is enabled in the reload event - Small cleanups... From moremanager - - * main/channel.c: Changes to manager events in channel.c - Newstate - event - Now has "CalleridNum" for numeric caller id, like - Newchannel - The event does not send "<unknown>" for unknown - caller IDs just an empty field - Newstate and Newchannel events - - these have changed headers "State" -> ChannelStateDesc Text based - channel state -> ChannelState Numeric channel state - The events - does not send "<unknown>" for unknown caller IDs just an empty - field - Newstate event - Now has "CalleridNum" for numeric caller - id, like Newchannel - The event does not send "<unknown>" for - unknown caller IDs just an empty field - Link and Unlink events - - The "Link" and "Unlink" bridge events in channel.c are now - renamed to "Bridge" - The link state is in the bridgestate: - header as "Link" or "Unlink" - For channel.c bridges, - "Bridgetype: core" is added. This opens up for bridge events in - rtp.c and channel drivers - The "Rename" manager event has a - renamed header, to use the same terminology for the current - channel as other events - Oldname -> Channel (Moremanager) - - * main/cdr.c: New manager event when a channel changes account - code. Maybe belongs in the new cdr category? ---moremanager--- - Event: NewAccountCode Modules: cdr.c Purpose: To report a change - in account code for a live channel Example: Event: NewAccountCode - Privilege: call,all Channel: SIP/olle-01844600 Uniqueid: - 1177530895.2 AccountCode: Stinas account 1234848484 - OldAccountCode: Olles Account 12345 - - * apps/app_dial.c: - Dial event - Event Dial has new headers, to - comply with other events - Source -> Channel Channel name - (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring - Dialstring in app data (moremanager) - - * apps/app_meetme.c: Adding small missing but important comma... - - * apps/app_meetme.c: A big oops... - - * apps/app_meetme.c: The MeetmeJoin now has caller ID name and - Caller ID number fields (like MeetMeLeave) (Moremanager) - - * channels/chan_zap.c: Update ZapShowChannels so that you can - specify one channel. Action ZapShowChannels Header changes - - Channel: -> ZapChannel For active channels, the Channel: and - Uniqueid: headers are added You can now add a "ZapChannel: " - argument to zapshowchannels actions to only get information about - one channel. From the moremanager branch - - * main/logger.c: Doxygen updates - - * include/asterisk/logger.h, /, main/logger.c, main/loader.c: - Merged revisions 91366 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4 - lines Make sure logger is reloaded at general reload in the cli. - (Discovered during Asterisk training in Portugal) ........ - - * main/manager.c: Change description of new manager command - - * main/manager.c, CHANGES: Add manager command for showing all - current channels. Thanks, eliel, for writing the original patch. - Modified by me to follow other manager events and the new - "moremanager" style. (closes issue #11478) Reported by: eliel - Patches: manager.c.patch uploaded by eliel (license 64) - -2007-12-06 04:37 +0000 [r91328] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Instead of iterating through the entire epoll - events array just look at the ones that will actually contain - data. (props to eliel on IRC for this) - -2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec - 2007) | 3 lines Reverting extra stuff I didn't mean to commit - ........ - - * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec - 2007) | 11 lines The 'G' option for Dial() did not properly - handle the case where only a label was provided. This was due to - the fact that the answering channel did not have an extension - set, so ast_parseable_goto would fail. This fix eliminates the - call to ast_parseable_goto on the answering channel since it is a - wasteful call. The answering channel and the calling channel are - both directed to the same extension and context, just different - priorities, so we can just copy the values from the calling - channel to the answering channel and increment the answering - channel's priority. (closes issue #11382, reported by jon, patch - by me with correction by jon) ........ - -2007-12-05 21:46 +0000 [r91238] Tilghman Lesher <tlesher@digium.com> - - * /, sounds/Makefile: Merged revisions 91237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007) - | 2 lines Upgrade to the latest version of extra sounds ........ - -2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant <russell@digium.com> - - * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) | - 10 lines Make the lock in the threadstorage debugging code - untracked to avoid a deadlock on thread destruction. (closes - issue #11207) Reported by: ys Patches: threadstorage.c.diff - uploaded by ys (license 281) Also fixes an open bug report: - (closes issue #11446) ........ - - * apps/app_directory.c: Resolve compiler warnings. - -2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, UPGRADE.txt, configs/manager.conf.sample, - CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change - cdr_manager to use a "CDR" level, rather than the (overcrowded) - "call" level. (Closes issue #11015) - - * CHANGES, apps/app_directory.c: Added multiple name listing. - (Closes issue #10413) - -2007-12-05 16:14 +0000 [r91171] Joshua Colp <jcolp@digium.com> - - * configs/http.conf.sample: Remove second prefix line. Only need it - documented once in the same file. (closes issue #11472) Reported - by: eserra Patches: http.conf.sample.diff uploaded by eserra - (license 45) - -2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename - "username" to "defaultuser" to match with "defaultip". "Username" - still works, but is deprecated. - - * channels/chan_sip.c: Remove the cseqs from "sip show channel" and - make more place for the call ID. - -2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: revert part of my changes from earlier today - since this code is no longer dependent on libpri.h - -2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant <russell@digium.com> - - * res/res_odbc.c: Use ast_free() instead of free(). (closes issue - #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded - by Laureano (license 265) - - * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04 - Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were - reported as happening on Mac systems. It turns out that the - problem was the Mac version of the ast_atomic_fetchadd_int() - function. The Mac atomic add function returns the _new_ value, - while this function is supposed to return the old value. So, the - crashes happened on unreferencing objects. If the reference count - was decreased to 1, ao2_ref() thought that it had been decreased - to zero, and called the destructor. However, there was still an - outstanding reference around. (closes issue #11176) (closes issue - #11289) ........ - - * /, main/utils.c: Merged revisions 91074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | - 4 lines When DEBUG_THREADS is enabled, we only have the details - about who is holding a lock that we are waiting on for a mutex, - not rwlocks. This should fix the problem where people have - reported "core show locks" crashing sometimes. ........ - - * channels/chan_zap.c: Fix mwimonitornotify on reload ... again. - This option was only read at startup so a reload would erase it - and not reset it. (pointed out by tzafrir) - - * utils/astman.c: Fix the build of astman. Any file that includes - any asterisk sub-headers needs to first include asterisk.h. - (closes issue #11394) - -2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Don't error when we don't have libpri - installed with libss7 support. Also, print the debug message - anyway if we can't find the right PRI - -2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 90967 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | - 7 lines Make some changes to some additions I made recently for - doing channel autoservice when looking up extensions. This code - was added to handle the case where a dialplan switch was in use - that could block for a long time. However, the way that I added - it, it did this for all extension lookups. However, lookups in - the in-memory tree of extensions should _not_ take long enough to - matter. So, move the autoservice stuff to be only around - executing a switch. ........ - - * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I - guess I only added this line in my head. (thanks to tzafrir for - pointing it out) - -2007-12-04 21:46 +0000 [r90993] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue - #11412) - -2007-12-04 21:23 +0000 [r90991] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c, CHANGES: Add manager action - 'sipshowregistry'. Closes issue #11464, patch by eliel. - -2007-12-04 19:08 +0000 [r90949] Russell Bryant <russell@digium.com> - - * include/asterisk/callerid.h, channels/chan_zap.c, - main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support - for monitoring MWI on FXO lines. This introduces two new options - for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor - option enables MWI monitoring. When the MWI state on a line - changes, then the script specified by mwimonitornotify will be - executed for custom handling of the state change, similar to the - externnotify option of voicemail.conf. Also, when the MWI state - on an FXO line changes, an internal Asterisk event is generated - to indicate the new state of the associated mailbox. That may, - any module that cares about MWI information will get notified and - can handle it just as if app_voicemail had sent this - notification. (BE-253, original patch from markster, with some - minor modifications by me to add comments, documentation, and - internal event support) - -2007-12-04 18:29 +0000 [r90930] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Kevin suggested doing the reverse of my - last commit, since imap_retrieve_file does not modify the - contents of the "mailbox" string. In other words, I'm changing - the imap_retrieve_file function to take a const char* as the - third argument so that I don't need to cast const char*'s as - char*'s to suppress compiler warnings. - -2007-12-04 18:15 +0000 [r90929] Jason Parker <jparker@digium.com> - - * Makefile: Add Makefile alias target 'pdf' which does the same - thing as asterisk.pdf. Issue 11452, reported by blitzrage. - -2007-12-04 18:14 +0000 [r90928] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Suppress a compiler warning due to - discarding a "const" qualifier - -2007-12-04 18:09 +0000 [r90927] Jason Parker <jparker@digium.com> - - * main/global_datastores.c: Fix build, that some people aren't - seeing for some reason. - -2007-12-04 17:51 +0000 [r90899] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Wrong locking style got merged from 1.4 to - trunk. My mistake. - -2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: fix build of this module when libpri and/or - libss7 are or are not present - -2007-12-04 17:38 +0000 [r90879] Jason Parker <jparker@digium.com> - - * main/channel.c, /: Merged revisions 90876 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600 - (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after - allocating a timing fd, we need to make sure to close it. Issue - 11454, patch by eliel. ........ - -2007-12-04 17:36 +0000 [r90878] Russell Bryant <russell@digium.com> - - * main/Makefile: Fix a silly little typo :) - -2007-12-04 17:35 +0000 [r90877] Jason Parker <jparker@digium.com> - - * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but - blocked in trunk since this hadn't been merged yet. - -2007-12-04 17:08 +0000 [r90873] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, main/global_datastores.c (added), - channels/chan_local.c, /, main/Makefile, - include/asterisk/channel.h, include/asterisk/global_datastores.h - (added), apps/app_queue.c: Merged revisions 90735 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 - Dec 2007) | 22 lines A big one... This is the merge of the - forward-loop branch. The main change here is that call-forwards - can no longer loop. This is accomplished by creating a datastore - on the calling channel which has a linked list of all devices - dialed. If a forward happens, then the local channel which is - created inherits the datastore. If, through this progression of - forwards and datastore inheritance, a device is attempted to be - dialed a second time, it will simply be skipped and a warning - message will be printed to the CLI. After the dialing has been - completed, the datastore is detached from the channel and - destroyed. This change also introduces some side effects to the - code which I shall enumerate here: 1. Datastore inheritance has - been backported from trunk into 1.4 2. A large chunk of code has - been removed from app_dial. This chunk is the section of code - which handles the call forward case after the channel has been - requested but before it has been called. This was removed because - call-forwarding still works fine without it, it makes the code - less error-prone should it need changing, and it made this set of - changes much less painful to just have the forwarding handled in - one place in each module. 3. Two new files, global_datastores.h - and .c have been added. These are necessary since the datastore - which is attached to the channel may be created and attached in - either app_dial or app_queue, so they need a common place to find - the datastore info. This approach was taken in case similar - datastores are needed in the future, there will be a common place - to add them. ........ - -2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson <oej@edvina.net> - - * apps/app_queue.c: (closes issue #11431) Reported by: Laureano - Patches: app_queue.c.patch uploaded by Laureano (license 265) - - * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel - Patches: core.show.hint.patch uploaded by eliel (license 64) - - * CHANGES: (closes issue #11462) Reported by: eliel Patches: - CHANGES.patch uploaded by eliel (license 64) - -2007-12-04 15:01 +0000 [r90851] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Pass the Asterisk version to AGI scripts as part - of the initial dump of info Reported by: acunningham Patch by: - acunningham (Closes issue #11398) - -2007-12-04 11:50 +0000 [r90834] Luigi Rizzo <rizzo@icir.org> - - * res/Makefile: fix build on cygwin - -2007-12-03 23:52 +0000 [r90760] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 - Dec 2007) | 5 lines Solaris requires the inclusion of - sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: - snuffy,tilghman (Closes issue #11430) ........ - -2007-12-03 23:49 +0000 [r90746] Steve Murphy <murf@digium.com> - - * main/hashtab.c: A small fix from snuffy - -2007-12-03 23:48 +0000 [r90738] Jason Parker <jparker@digium.com> - - * res/res_monitor.c: Add manager events for when a monitor is - started or stopped. Closes issue #10191, patch by dgradecak. - -2007-12-03 23:29 +0000 [r90737] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 - Dec 2007) | 5 lines If both dbhost and dbsock were not set, a - NULL deref could result Reported by: xrg Patch by: tilghman - (Closes issue #11387) ........ - -2007-12-03 22:07 +0000 [r90697] Jason Parker <jparker@digium.com> - - * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600 - (Mon, 03 Dec 2007) | 4 lines Make sure we always close the - conference fd if we have an open one. Issue 11383, reported by - markmhy, patch by eliel. ........ - -2007-12-03 21:24 +0000 [r90670] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Replacing some calls to free() with - ast_free(). (closes issue #11448, reported and patched by jaroth) - -2007-12-03 21:03 +0000 [r90656] Joshua Colp <jcolp@digium.com> - - * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands - for speech recognition. These mirror the dialplan applications - mostly but present the information in a nicer fashion. The SPEECH - RECOGNIZE command for example will return the results instead of - having to query the dialplan functions. - -2007-12-03 21:00 +0000 [r90644] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec - 2007) | 5 lines Changing some bad logic when calculating the - interdigit timeout. (closes issue #11402, reported and patched by - eferro) ........ - -2007-12-03 20:58 +0000 [r90631] Jason Parker <jparker@digium.com> - - * /, res/res_features.c: Merged revisions 90607 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600 - (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce - application. Issue #11436, reported by lytledd, patch by eliel. - ........ - -2007-12-03 20:30 +0000 [r90591] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c: Avoid an additional function call. Reported by: - eliel Patch by: eliel (Closes issue #11438) - -2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 90588 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 - lines Do not create a smoother for G723.1 frames, they need to be - left alone to their native 20/24 byte size. ........ - - * main/channel.c, /, include/asterisk/channel.h, .cleancount: - Merged revisions 90548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 - lines Preserve the indication currently playing on a channel when - a masquerade operation happens. (issue #BE-88) ........ - -2007-12-03 16:46 +0000 [r90528] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample: Updating sample queues.conf file to - show how multiple periodic announcements may be specified since - this was not documented previously (closes issue #11432, reported - and patched by Laureano) - -2007-12-03 14:14 +0000 [r90508] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Remove the file descriptors from the main poll - channel when the channel is hung up during the dialing attempt, - and make sure a channel exists before trying to remove it at the - end. (closes issue #11441) Reported by: blitzrage - -2007-12-02 18:20 +0000 [r90471] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | - 6 lines The other day when I went through making changes as a - result of the ao2_link() change, I added some code to set - pointers to NULL after they were unreferenced. This pointed out - that in this place, the object was unreferenced before the code - was done using it. So, move the unref down a little bit. (crash - reported by jmls on IRC) ........ - -2007-12-02 09:42 +0000 [r90433] Tilghman Lesher <tlesher@digium.com> - - * main/autoservice.c, /: Merged revisions 90432 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) - | 7 lines Clarify the return value on autoservice. Specifically, - if you started autoservice and autoservice was already on, it - would erroneously return an error. Reported by: adiemus Patch by: - dimas (Closes issue #11433) ........ - -2007-12-01 01:37 +0000 [r90410] Jason Parker <jparker@digium.com> - - * res/res_adsi.c: Only reload if the config file has changed. - Closes issue #11281, patch by eliel. - -2007-11-30 21:19 +0000 [r90388] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, include/asterisk/app.h, - include/asterisk/audiohook.h, res/res_features.c, - include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c, - configs/features.conf.sample: Adding support for the - "automixmonitor" dial and queue options. This works in much the - same way as the automonitor, except that instead of using the - monitor app, it uses the mixmonitor app. By providing an 'x' or - 'X' as a dial or queue option, a DTMF sequence may be entered (as - defined in features.conf) to start the one-touch mixmonitor. This - patch also introduces some new API calls to the audiohooks code - for searching for an audiohook by type and for searching for a - running audiohook by type. Big thanks to joetester for writing - the initial patch, testing it and patiently waiting for it to be - committed. (closes issue #10185, reported and patched by - xmarksthespot) - -2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant <russell@digium.com> - - * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c, - channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged - revisions 90348 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | - 8 lines Change the behavior of ao2_link(). Previously, in - inherited a reference. Now, it automatically increases the - reference count to reflect the reference that is now held by the - container. This was done to be more consistent with ao2_unlink(), - which automatically releases the reference held by the container. - It also makes it so it is no longer possible for a pointer to be - invalid after ao2_link() returns. ........ - - * /, include/asterisk/astobj2.h: Merged revisions 90310 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) | - 2 lines Add some notes on the behavior of ao2_unlink() after a - discussion with Tilghman ........ - -2007-11-30 14:45 +0000 [r90270] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 - lines Fix locking issues under one legged replaces scenarios. - (closes issue #11420) Reported by: irroot Patches: - chan_sip_oneleg.patch uploaded by irroot (license 52) ........ - -2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov - 2007) | 5 lines Clear the DTMF buffer if the call times out. - (closes issue #11418, reported and patched by eferro) ........ - - * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov - 2007) | 6 lines This patch handles the case where a queue member - with a negative penalty is added via the manager. If a negative - value is submitted for a member penalty, we set it to 0. (closes - issue #11411, reported and patched by Laureano) ........ - -2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29 - Nov 2007) | 2 lines Properly escape input buffers (Fixes - AST-2007-025) ........ - - * /, formats/format_wav.c, formats/format_pcm.c, - formats/format_ogg_vorbis.c, main/file.c, - include/asterisk/mod_format.h, formats/format_h263.c, - formats/format_h264.c, formats/format_wav_gsm.c, - formats/format_g726.c: Merged revisions 90155 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) - | 5 lines Use of "private" as a field name in a header file - messes with C++ projects Reported by: chewbacca Patch by: casper - (Closes issue #11401) ........ - - * include/asterisk/lock.h: Fix build of trunk - - * /, sounds/Makefile: Merged revisions 90154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) - | 2 lines Upgrade the core sounds release version ........ - -2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming <kpfleming@digium.com> - - * utils/Makefile, utils/hashtest.c: let's try this again... *all* - compilation and linking in Asterisk should be done using the - standard compilation rules, not manually created ones. changing - hashtest.c to use these rules caused the compiler to notice a - large number of coding guidelines violations, so those are fixed - too. - - * main/manager.c: restore behavior from the 1.4 branch... manager - users created via users.conf should default to *all* permissions, - not none - -2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant <russell@digium.com> - - * main/channel.c, /, include/asterisk/channel.h, - funcs/func_callerid.c: Merged revisions 90145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | - 5 lines This set of changes is to make some callerID handling - thread-safe. The ast_set_callerid() function needed to lock the - channel. Also, the handlers for the CALLERID() dialplan function - needed to lock the channel when reading or writing callerid - values directly on the channel structure. ........ - - * include/asterisk/file.h, /, main/file.c: Merged revisions 90142 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | - 4 lines Merge a change from team/russell/chan_refcount ... This - makes ast_stopstream() thread-safe. ........ - - * include/asterisk/audiohook.h: Merge another small doxygen change - from team/russell/chan_refcount to indicate that a channel - doesn't need to be locked before calling a certain function. - - * include/asterisk/channel.h: Merge some channel.h doxygen updates - from team/russell/chan_refcount This was mostly to note whether a - channel needed to be locked or not before calling these - functions. However, I added some other things, too. - -2007-11-28 23:03 +0000 [r90102] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions - 90101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 - lines Fix a few memory leaks. (closes issue #11405) Reported by: - eliel Patches: load_realtime.patch uploaded by eliel (license 64) - ........ - -2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming <kpfleming@digium.com> - - * configs/users.conf.sample, main/manager.c, /: Merged revisions - 90098 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) - | 2 lines it is impossible to set permissions for manager - accounts created by users.conf (reported internally, patched by - me) ........ - -2007-11-28 22:32 +0000 [r90099] Joshua Colp <jcolp@digium.com> - - * main/cli.c: file says... compile before you commit! - -2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: Removing a pointless check of option_debug - - * main/pbx.c, /: Merged revisions 90059 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov - 2007) | 13 lines Removing some seemingly pointless code. This - sets a channel variable for every priority executed in the - dialplan if you have debug set to anything non-zero. This seems - pointless due to the fact that these channel variables are not - referenced anywhere else in the code and their names are esoteric - enough that they would not be practical to reference in the - dialplan. Plus the fact that this behavior isn't documented - anywhere means that the change is not likely to cause any - disruption. If anything, this may actually cause a slight - performance increase if running with debug on. The motivating - influence for this code change is the eventwhencalled option for - queues. If set to vars, all channel variables will be output to - the manager. These unnecessary channel variables make the output - a lot more difficult to deal with. ........ - -2007-11-28 20:33 +0000 [r90039] Steve Murphy <murf@digium.com> - - * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit - transparent - -2007-11-28 20:27 +0000 [r90038] Jason Parker <jparker@digium.com> - - * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c: - Remove "old"-style CLI handler, since nothing uses it anymore. - Closes issue #11403, patch by eliel. This also completes the - janitor project. - -2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp <jcolp@digium.com> - - * main/cli.c: Hide CLI commands starting with _ from tab completion - as was done previously. (closes issue #11395) Reported by: eliel - Patches: cli.c.patch uploaded by eliel (license 64) - - * main/abstract_jb.c, res/res_agi.c: Fix a few log messages. - (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff - uploaded by IgorG (license 20) - -2007-11-28 00:49 +0000 [r89947] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Merge some little changes from - team/russell/chan_refcount to help reduce the diff to trunk. This - just removes some checks on the return value of alloca(), as - behavior is undefined if it runs out of stack space, and we don't - check it anywhere else. - -2007-11-28 00:47 +0000 [r89946] Mark Michelson <mmichelson@digium.com> - - * configs/musiconhold.conf.sample, configs/extconfig.conf.sample, - res/res_musiconhold.c, CHANGES: Adding support for realtime music - on hold. The following are the main points: 1. When moh is - started, we search first in memory to find the class. If we do - not find it in memory, we search realtime instead. 2. When moh is - restarted (as in, it had been started on this particular channel, - stopped, and now we're starting it again), if using the "files" - mode, then realtime will always be rechecked. If you are using - other modes, however, we will simply reattach to the external - running process which was playing moh earlier in the call. This - is a necessary compromise so that we don't end up with too many - background processes. 3. musiconhold.conf has a general section - now. It has one option: cachertclasses. If set to yes, then moh - classes found in realtime will be added to the in-memory list. - This has the advantage of not requiring database lookups each - time moh is started, but it has the disadvantage of not truly - being realtime. I have tested this for functionality, and it - passes. I also tested this under valgrind and there are no memory - problems reported under typical use. Special thanks to Sergee for - implementing this feature and enduring my complaints on the - bugtracker! (closes issue #11196, reported and patched by sergee) - -2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant <russell@digium.com> - - * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | - 4 lines - update documentation for some of the goto functions to - note that they handle locking the channel as needed - update - ast_explicit_goto() to lock the channel as needed ........ - - * include/asterisk/channel.h: Document that the channel is not - locked when the send_digit_begin and end callbacks get called. - - * main/autoservice.c, /: Merged revisions 89886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | - 2 lines Don't do frame processing if ast_read() returned NULL. - ........ - - * channels/chan_iax2.c: Merge changes from - team/russell/iax2_frame_queue This patch is an optimization for - chan_iax2. This module is now heavily multi-threaded. However, - there is still a good number of globally shared resources that - prevent things from happen asynchronously. One of those things - was the global IAX frame queue. This queue was used to hold - frames that have been deferred for transmitting by another - thread, and frames that may need to get retransmitted. I changed - the frame queue to be per-call, since almost all of the frame - queue handling only cares about frames specific to a call number. - - * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | - 3 lines Instead of depending on the return value of ast_true(), - explicitly set the eventwhencalled variable to 1. ........ - - * main/pbx.c, /: Merged revisions 89839 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | - 2 lines Don't start/stop autoservice in pbx_extension_helper() - unless a channel exists ........ - -2007-11-27 23:11 +0000 [r89838] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov - 2007) | 12 lines Two changes with regards to the - 'eventwhencalled' option of queues.conf 1) Due to some signed vs. - unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' - did exactly the same thing. Thus the sign change of the ast_true - call. 2) The vars2manager function overwrote a \n for every - channel variable it parsed, resulting in bizarre output for the - channel variables. This patch remedies this. (related to issue - #11385, however I'm not sure if this will actually be enough to - close it) ........ - -2007-11-27 22:42 +0000 [r89835] Russell Bryant <russell@digium.com> - - * channels/chan_misdn.c: Bring in a small change from - team/russell/chan_refcount This replaces tab completion code with - the use of a public function that does the same thing - -2007-11-27 22:14 +0000 [r89792] Steve Murphy <murf@digium.com> - - * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the - conditional unlock of the contexts when the hash table is used - instead; also, used the ast_free_ptr as advised. - -2007-11-27 22:05 +0000 [r89791] Russell Bryant <russell@digium.com> - - * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | - 41 lines Merge changes from team/russell/autoservice_1.4 This set - of changes fixes an issue that was reported to me on IRC - yesterday. The user, d1mas, was using chan_zap for incoming calls - and was having DTMF recognition issues in some situations. - Specifically, he noticed that the problem occurred when using - DISA or WaitExten. He also noticed that when using Read, the - problem did not occur. His system also used DUNDi for dialplan - lookups. So, he theorized that if the DUNDi lookups blocked for - some period of time, that audio from the zap channel could get - lost. If the audio got lost, then it wouldn't be run through the - DTMF detector, and digits could get lost. He was correct, and the - following set of changes fixes the problem. However, the changes - go a little bit further than what was necessary to fix this exact - problem. 1) I updated pbx_extension_helper() to autoservice the - associated channel to handle cases where extension lookups may - take a long time. This would normally be a dialplan switch that - does some lookup over the network, such as the DUNDi or IAX2 - switches. This ensures that even while a DUNDi lookup is - blocking, the channel will be continuously serviced. 2) I made a - change to the autoservice code. This is actually something that - has bothered me for a long time. When a channel is in - autoservice, _all_ frames get thrown away. However, some frames - really shouldn't be thrown away. The most notable examples are - signalling (CONTROL) frames, and DTMF. So, this patch queues up - important frames while a channel is in autoservice. When - autoservice is stopped on the channel, the queued up frames get - stuck back on the channel so that they can get processed instead - of thrown away. 3) I made another change to the autoservice code - to handle the case where autoservice is started on channels - recursively. Previously, you could call ast_autoservice_start() - multiple times on a channel, and it would stop the first time - ast_autoservice_stop() gets called. Now, it will ensure that - autoservice doesn't actually stop until the final call to - ast_autoservice_stop(). ........ - -2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson <oej@edvina.net> - - * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A - few more "moremanager" fixes - - * include/asterisk.h, main/asterisk.c, main/loader.c: More - "moremanager" fixes. Manager commands to check module status. - - * include/asterisk/manager.h: More "moremanager" changes - doxygen - docs and changing manager version (finally) before making more - dramatic changes. - - * channels/chan_iax2.c: More additions from the "moremanager" - branch, this time for IAX2. - -2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/app.c: Merged revisions 89709 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) - | 2 lines on second thought... revert all the other changes i've - made in app options parsing leaving only one: if an empty - argument is supplied for an option, set that argument pointer to - point to an empty string rather than NULL, so that the - application can do normal checks on it without worrying about it - being NULL ........ - -2007-11-27 20:17 +0000 [r89710] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: remove a duplicate manager event - -2007-11-27 19:50 +0000 [r89706] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Manager events from the "moremanager" - branch - -2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/app.c: Merged revisions 89701 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) - | 2 lines generate a warning when an application option that - requires an argument is ignored due to lack of an argument - ........ - -2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Starting to merge changes from the - "moremanager" branch. Documentation will follow. - - * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The - following patch with updates for trunk. Works much better in - trunk. Also by accident fixed a bad typo by a previous committer, - which actually made video calls not work fully... Merged - revisions 89630 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 - lines If we get a codec offer using a well-known payload type, - but using it for another codec that we don't know, Asterisk did - not remove that codec from the list. With this patch, we remove - the codec from audio and video rtp objects and deny it ever - existed. Thanks to lasse for testing. (closes issue #11376) - Reported by: lasse Patches: bug11376.txt uploaded by oej (license - 306) Tested by: lasse ........ - -2007-11-27 19:12 +0000 [r89683] Jason Parker <jparker@digium.com> - - * include/asterisk/strings.h: Add an S_COR macro, which is similar - to the existing S_OR macro, except with an additional boolean - arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: - S_COR(foo, bar, "baz") - -2007-11-27 18:50 +0000 [r89682] Steve Murphy <murf@digium.com> - - * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, - pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, - pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, - pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22, - res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3, - pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, - pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex, - pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8: - made AEL 8-bit transparent; mainly the lexer was tossing chars - with the hi-order bit set. Not nice. Also, allow @ in extension - names, and a backslash, also. - -2007-11-27 17:01 +0000 [r89637] Joshua Colp <jcolp@digium.com> - - * main/utils.c: Ensure the value returned from ast_random is - between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348) - Reported by: sperreault - -2007-11-27 16:13 +0000 [r89635] Russell Bryant <russell@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 89634 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | - 3 lines Add a note to the sample voicemail config noting that - when using IMAP storage, only the first format specified will be - attached to the message. ........ - -2007-11-27 15:41 +0000 [r89632] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) - | 3 lines Default result of STAT should be "0" not "". Reported - via the -users mailing list, fixed by me. ........ - -2007-11-27 07:36 +0000 [r89625] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov - 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) - Reported by: pj ........ - -2007-11-27 06:47 +0000 [r89623] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample, - include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 - line closes issue #11379; OK, this is an attempt to make both - sides happy. To the cdr.conf file, I added the option - 'unanswered', which defaults to 'no'. In this mode, you will see - a cdr for a call, whether it was answered or not. The disposition - will be NO ANSWER or ANSWERED, as appropriate. The src is as - you'd expect, the destination channel will be one of the channels - from the Dial() call, usually the last in the list if more than - one chan was specified. With unanswered set to 'yes', you will - still see this cdr entry in both cases. But in the case where the - dial timed out, you will also see a cdr for each line attempted, - marked NO ANSWER, with no destination channel name. The new - option defaults to 'no', so you don't see the pesky extra cdr's - by default, and you will not see the irritating 'not posted' - messages. ........ - -2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson <mmichelson@digium.com> - - * pbx/ael/ael-test/ael-test19/extensions.ael, - pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt, - pbx/ael/ael-test/ael-test3/extensions.ael, - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances - of "CALLERID(number)" to "CALLERID(num)" for consistency's sake - (closes issue #11381, reported and patched by jon) - - * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov - 2007) | 7 lines After issuing a "say load new", if a caller hangs - up during the middle of playback of a number, app_playback will - continue to try to play the remaining files. With this change, no - more files will be played back upon hangup. (closes issue #11345, - reported and patched by IgorG) ........ - -2007-11-26 22:52 +0000 [r89615] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the configure script check for - libpri to check for the newest function that was just added. - Cresl1n, please keep this in mind when making these changes to - libpri or libss7. - -2007-11-26 21:23 +0000 [r89613] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Rename - "limitonpeer" to "counteronpeer" since the call-limit is - deprecated. Both still works in this version. - -2007-11-26 21:14 +0000 [r89612] Joshua Colp <jcolp@digium.com> - - * main/dial.c, /: Merged revisions 89610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 - lines Fix issues with async dialing with an application - executing. The application has to be terminated and control - returned to the thread before hanging things up. (issue #BE-252) - ........ - -2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Formatting, doxygenification - - * channels/chan_sip.c: Formatting changes, cleaning up some code - - * include/asterisk/doxyref.h, channels/chan_sip.c: Start using - Doxygen groupings to group variables and defines. - - * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark - "concise" as deprecated - Restructure other changes to - UPGRADE.txt and CHANGES We're still looking for scripts that - replace asterisk -rx "show shannels concise" by using the manager - interface, but still produces the same output. Anyone? - -2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp <jcolp@digium.com> - - * res/res_features.c, apps/app_queue.c: Perform some module use - counting audits. This is now done outside the scope of the - application/dialplan function so they do not need to worry about - it. - - * /, res/res_features.c: Merged revisions 89599 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 - lines Add module counting removal for error conditions. (closes - issue #11333) Reported by: Laureano Patches: - res_features_v2.c.patch uploaded by Laureano (license 265) - ........ - -2007-11-26 17:49 +0000 [r89596] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 89594 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | - 3 lines Add channel locking to a function that needed to be doing - it. This is just a little something I noticed while working on a - completely unrelated issue. ........ - -2007-11-26 17:46 +0000 [r89595] Steve Murphy <murf@digium.com> - - * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes - issue #11341; made changes to make utils again right with the - MTX_PROFILE world. - -2007-11-26 17:38 +0000 [r89593] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 - lines Use ast_free to free memory, or else we shall implode if - MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys - Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) - ........ - -2007-11-26 17:26 +0000 [r89591] Steve Murphy <murf@digium.com> - - * main/hashtab.c: closes issue #11356; Many thanks to snuffy for - his code review and changes to cut down duplication. I tested - this against hashtest, and it passes. I reviewed the changes, and - they look reasonable. I had to remove a few const decls to make - things compile on my workstation, - -2007-11-26 17:25 +0000 [r89590] Russell Bryant <russell@digium.com> - - * Makefile: make sure we check to see if the configure script has - been executed on a new checkout or after a distclean - -2007-11-26 17:23 +0000 [r89589] Joshua Colp <jcolp@digium.com> - - * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov - 2007) | 6 lines Close the audio file before sending it to the - post processing application. (closes issue #11357) Reported by: - reformed Patches: mixmonitor.patch uploaded by reformed (license - 330) ........ - -2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) - | 2 lines when parsing application options that take arguments, - don't indicate that the option was supplied unless a - non-zero-length argument was found for it ........ - -2007-11-26 16:24 +0000 [r89583] Steve Murphy <murf@digium.com> - - * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to - pnlarsson for noting the spelling error in the cli commands. - Also, added some verbage about the new algorithm to CHANGES. - -2007-11-26 16:20 +0000 [r89582] Joshua Colp <jcolp@digium.com> - - * main/utils.c: Revert change for 11348 until it can be looked at - even more. - -2007-11-26 15:50 +0000 [r89581] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov - 2007) | 6 lines Revert vmu->email back to an empty string if it - was empty when imap_store_file was called. This prevents sending - a duplicate e-mail. (closes issue #11204, reported by spditner, - patched by me) ........ - -2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 89577 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 - lines If channel allocation fails because the alert pipe could - not be created also free the scheduler context. (closes issue - #11355) Reported by: eliel Patches: main.channel.c.patch uploaded - by eliel (license 64) ........ - - * main/utils.c: Make the behavior of using /dev/urandom for random - numbers the same as random(). (closes issue #11348) Reported by: - sperreault Patches: ast_random2.diff uploaded by sperreault - (license 252) - - * channels/chan_sip.c: Instead of printing out one codec in sip - show channels print out all of the native ones (this is for - video). (closes issue #11366) Reported by: ovi - - * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 - lines When unloading app_meetme destroy any auto created contexts - created by SLA. (closes issue #11367) Reported by: eliel ........ - - * apps/app_controlplayback.c: Don't crash if the 'o' option of - ControlPlayback is used without any value. (closes issue #11375) - Reported by: johan - -2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson <oej@edvina.net> - - * channels/chan_usbradio.c: Formatting changes - - * main/channel.c, include/asterisk/channel.h: Try to get channel.h - and channel.c aligned in regards to ast_set_callerid as well as - change name of variables to follow the rest of the naming. - -2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, res/res_config_odbc.c, /, - res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions - 89559 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) - | 14 lines We previously attempted to use the ESCAPE clause to - set the escape delimiter to a backslash. Unfortunately, this does - not universally work on all databases, since on databases which - natively use the backslash as a delimiter, the backslash itself - needs to be delimited, but on other databases that have no - delimiter, backslashing the backslash causes an error. So the - only solution that I can come up with is to create an option in - res_odbc that explicitly specifies whether or not backslash is a - native delimiter. If it is, we use it natively; if not, we use - the ESCAPE clause to make it one. Reported by: elguero Patch by: - tilghman (Closes issue #11364) ........ - - * channels/chan_sip.c: Typo (someone needs to test compile before - committing his changes) - -2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: More doxygen changes - - * channels/chan_sip.c: Housekeeping - - * channels/chan_sip.c: Formatting, doxygen updates - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: - - Deprecate "call-limit" in chan_sip. No other channel driver - enforces call-limits and we now have the groupcount system to - implement call-limits in the dialplan. You can use the "setvar" - option in realtime/sip.conf to set limits per device. - Implement - "callcounter" as a new option to enable the call counting we need - to report device status to queue, manager and SIP subscriptions. - The call counter setting is now enabled in the code by setting - the device call-limit to 999. When we remove the call limit, we - can simply enable this with a boolean setting. - - * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping... - - Fix typo in chan_sip - Remove changes to caller ID structure, - moving it to branch (russellb) - -2007-11-24 21:00 +0000 [r89547] Steve Murphy <murf@digium.com> - - * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c, - configs/extensions.conf.sample: closes issue #11363; where the - pattern _20x. buried in an included context, didn't match 2012; - There were a small set of problems to fix: 1. I needed NOT to - score patterns unless you are at the end of the data string. 2. - Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize - the patterns in the trie to caps. 3. When a pattern ends with dot - or exclamation, CANMATCH/MATCHMORE should always report this - pattern, no matter the length. With this commit, I also supplied - the wish of Luigi, where the user can select which pattern - matching algorithm to use, the old (legacy) pattern matcher, or - the new, trie based matcher. The OLD matcher is the default. A - new [general] section variable, extenpatternmatchnew, is added to - the extensions.conf, and the example config has it set to false. - If true, the new matcher is used. In all other respects, the - context/exten structs are the same; the tries and hashtabs are - formed, but in the new mode the tries are not used. A new CLI - command 'dialplan set extenpatternmatch true/false' is provided - to allow switching at run time. I beg users that are forced to - return to the old matcher to please report the reason in the bug - tracker. Measured the speed benefit of the new matcher against an - impossibly large context with 10,000 extensions: the new matcher - is 374 times faster. - -2007-11-24 17:07 +0000 [r89546] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007) - | 5 lines Free some frames that would otherwise leak on error. - Reported by: Laureano Patch by: Laureano,tilghman (Closes issue - #11351) ........ - -2007-11-24 16:53 +0000 [r89544] Steve Murphy <murf@digium.com> - - * main/app.c: Added <sys/file.h> include to allow trunk to compile. - Hope this doesn't louse thing up. - -2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_h323.c: remove a DEBUG_THREADS message that - accesses private lock fields. If needed, the code to extract this - information should be implemented in some generic header or - library and the function called here. (closed bug #11362) - - * main/acl.c, main/http.c, main/app.c: remove some unnecessary - includes - -2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) - | 9 lines Currently, zero-length voicemail messages cause a - hangup in VoicemailMain. This change fixes the problem, with a - multi-faceted approach. First, we do our best to avoid these - messages from being created in the first place, and second, if - that fails, we detect when the voicemail message is zero-length - and avoid exiting at that point. Reported by: dtyoo Patch by: - gkloepfer,tilghman (Closes issue #11083) ........ - - * main/manager.c, /: Merged revisions 89536 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) - | 10 lines Up until this point, the XML output of the manager has - been technically invalid, due to the repetition of certain - parameters in a single event. This caused various issues for XML - parsers, some of which refused to parse at all, given the - invalidity of the rendered XML. So this commit fixes the XML - output, ensuring that each entity parameter has a unique name, - thus ensuring valid XML. Reported by: msetim Patch by: tilghman - (Closes issue #10220) ........ - - * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23 - Nov 2007) | 5 lines Use ESCAPE clause for the first parameter, - not just 2nd-Nth parameters. Reported by: apsaras Patch by: - tilghman (Closes issue #11353) ........ - -2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: put in the necessary hooks for video support - in the console. This is a NOP as far as the current code is - concerned, but there is already support in ./configure and the - Makefiles for the various libraries used by console_video.c (not - yet in the tree) so addition is trivial. - - * channels/chan_sip.c: set rtpmap video info according to what is - read from SDP; make the format explicit in a debug message; print - the audio instead of aggregated peer capability in a debugging - msg. - -2007-11-23 09:40 +0000 [r89531] Olle Johansson <oej@edvina.net> - - * include/asterisk/channel.h: Let's start with implementing the - base architecture for UTF8 caller ID's so we can handle multiple - formats properly. This is not carved in stone, but a proposal to - start with. We need to add support for transliterations as well - as UTF8 handling, propably with libiconv. Murf is looking into - that for the dialplan. - -2007-11-23 09:03 +0000 [r89530] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/image.h, formats/format_jpeg.c: formatting - cleanup on the header, normalization of the assignment of - descriptor fields. - -2007-11-23 02:37 +0000 [r89529] Russell Bryant <russell@digium.com> - - * configs/agents.conf.sample, /: Merged revisions 89527 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) | - 3 lines mvanbaak pointed out a spelling error in this sample - configuration file. While I was at it, I went ahead and tweaked - it a little bit more. ........ - -2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo <rizzo@icir.org> - - * doc/CODING-GUIDELINES: new info on the management of headers - - * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c, - apps/app_milliwatt.c: more header removal - - * include/asterisk/channel.h: formatting cleanup - - * include/asterisk.h, apps/app_read.c, apps/app_record.c, - apps/app_echo.c, apps/app_readexten.c, - include/asterisk/channel.h, apps/app_system.c, - apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h, - apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c, - include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c, - apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c, - apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a - little bit the content of header files to reduce dependencies. In - this commit: - move the ast_register/unregister_app functions to - module.h to avoid the need to include pbx.h for the simpler apps; - - move the ast_group structure to channel.h to remove the - dependency of app.h on linkedlists.h Note, this is a long process - that I am doing in small steps. The main difficulty is that now - for each subsystem we have a single header (e.g. channel.h) - included by the subsystem provider (usually one file, e.g. - channel.c) and by its clients (dozens of them, e.g. we have some - 70+ apps and 30+ functions). This requires the clients to include - all the extra headers required by the provider (eg. lock.h, - linkedlists.h, definitions of substructures...) even though many - of the clients would be just happy with opaque struct - declarations and function prototypes. The long term plan is to - eventually rectify this structure so that the compilation can - become faster, and also APIs are more stable. - - * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c, - apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove - some useless includes - - * include/asterisk/audiohook.h, apps/app_dictate.c, - apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c, - apps/app_mixmonitor.c, apps/app_stack.c, - apps/app_controlplayback.c: more removal of redundant headers - - * apps/app_read.c, apps/app_echo.c, apps/app_record.c, - apps/app_userevent.c, apps/app_image.c, apps/app_system.c, - apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c, - apps/app_while.c: remove redundant headers - - * main/file.c, main/netsock.c: more removal of fcntl.h and other - system headers - - * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, - codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c, - res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c, - formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c, - apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c, - codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which - are either useless or done elsewhere - - * formats/format_sln.c, formats/format_wav.c, - formats/format_ogg_vorbis.c, include/asterisk/_private.h, - formats/format_wav_gsm.c, formats/format_ilbc.c, - include/asterisk/file.h, formats/format_vox.c, - formats/format_pcm.c, main/file.c, formats/format_h263.c, - formats/format_g723.c, formats/format_h264.c, - include/asterisk/frame.h, formats/format_jpeg.c, - formats/format_g726.c, formats/format_gsm.c, - formats/format_g729.c: implement the split of file.h and - mod_format.h - - * include/asterisk/mod_format.h (added): Add a specific header for - providers of file and format handling routines, moving here - structs and function declarations formerly in file.h - -2007-11-21 23:54 +0000 [r89513] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c, - res/res_features.c, apps/app_queue.c, channels/chan_iax2.c: - closes issue #11285, where an unload of a module that creates a - dialplan context, causes a crash when you do a 'dialplan show' of - that context. This is because the registrar string is defined in - the module, and the stale pointer is traversed. The reporter - offered a patch that would always strdup the registrar string, - which is practical, but I preferred to destroy the created - contexts in each module where one is created. That seemed more - symmetric. There were only 6 place in asterisk where this is - done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, - and app_queue. The two apps destroyed the context, but left the - contexts. All is fixed now and unloads should be dialplan - friendly. - -2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo <rizzo@icir.org> - - * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c, - channels/chan_local.c, apps/app_record.c, funcs/func_strings.c, - apps/app_sayunixtime.c, apps/app_test.c, - apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, - apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, - channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c, - pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c, - apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c, - channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c, - apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c, - funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c, - channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c, - channels/iax2-provision.c, funcs/func_cut.c, - apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, - apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c, - apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, - channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, - pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, - apps/app_voicemail.c, channels/chan_unistim.c, - channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c, - apps/app_readexten.c, funcs/func_vmcount.c, - channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, - cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c, - channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c, - funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c, - apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, - apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c, - funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c, - apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, - apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c, - apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, - channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, - channels/chan_jingle.c, channels/chan_usbradio.c, - apps/app_channelredirect.c, apps/app_flash.c, - apps/app_directed_pickup.c, funcs/func_blacklist.c, - channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c, - channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c, - apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c: - remove another set of redundant #include "asterisk/options.h" - - * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c, - main/say.c, res/res_features.c, main/devicestate.c, main/utils.c, - res/res_musiconhold.c, res/res_jabber.c, main/indications.c, - main/enum.c, res/res_config_sqlite.c, main/config.c, - main/loader.c, main/term.c, main/cli.c, main/io.c, - main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c, - res/res_config_odbc.c, main/manager.c, res/res_agi.c, - main/http.c, main/logger.c, res/res_realtime.c, main/app.c, - main/image.c, main/dns.c, main/db.c, res/res_speech.c, - main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c, - main/translate.c, res/res_crypto.c, res/res_adsi.c, - main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c, - res/res_ael_share.c, res/res_monitor.c, main/rtp.c, - main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c, - main/adsistub.c, main/abstract_jb.c, main/file.c, - main/callerid.c, main/astmm.c, main/audiohook.c, - formats/format_g726.c, main/asterisk.c, res/res_odbc.c, - main/dsp.c: remove a bunch of useless #include "options.h" - -2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Remove unneccessary explicit case for BRI - - * channels/chan_zap.c: Take some debug code out :-) - -2007-11-21 22:20 +0000 [r89508] Luigi Rizzo <rizzo@icir.org> - - * main/cygload.c: add a missing return - -2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add BRI support to chan_zap - -2007-11-21 21:30 +0000 [r89506] Luigi Rizzo <rizzo@icir.org> - - * utils/Makefile, configure, configure.ac: enable support for stack - backtrace for stuff built in utils/ (this was present in the main - tree but forgotten here). - -2007-11-21 20:38 +0000 [r89505] Steve Murphy <murf@digium.com> - - * main/pbx.c: closes issue #11290; the proposed patch was a good - guess, and would solve the bug to some extent, but was really - masking the real issue, that there were bad entries in the table. - This fix removes the condition that the hashtab updates be done - on exten removal only when the pattern_tree was present, which is - silly. The operations that apply to the pattern tree are instead - made conditional. Also, threw back in routines that kpfleming - deleted because of probs in the 64-bit world. Tested on both 32 - and 64-bit machines (compile). Tested the reload problem with - over 20 reloads, and no problems. If you find more problems, - please reopen 11290. - -2007-11-21 20:22 +0000 [r89504] Terry Wilson <twilson@digium.com> - - * res/res_features.c: Simplify comparison in parking fix - -2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov - 2007) | 3 lines Fix a small error I made in my previous commit - ........ - - * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov - 2007) | 5 lines Changing an inaccurate debug message to be less - inaccurate. Under the circumstances, this message would always - report that there were 0 members available, even though that may - not be true. ........ - -2007-11-21 19:20 +0000 [r89492] Terry Wilson <twilson@digium.com> - - * /, res/res_features.c: Merged revisions 89491 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) | - 4 lines If a channel gets masqueraded in the middle of a park, - don't play the announcement to the masqueraded channel, and dial - back to the original channel on timeout. ........ - -2007-11-21 18:52 +0000 [r89490] Russell Bryant <russell@digium.com> - - * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove - the FAX_DETECT define and only do the calculations if fax - detection is enabled on the dsp. (closes issue #11331) Reported - by: dimas Patches: dsp.patch uploaded by dimas (license 88) - -2007-11-21 18:38 +0000 [r89489] Tilghman Lesher <tlesher@digium.com> - - * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set - READSTATUS as an indication of the result Also, some cleanup to - CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman - (Closes issue #11004) - -2007-11-21 18:24 +0000 [r89488] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix a small gramatical error in a comment - -2007-11-21 18:19 +0000 [r89487] Mark Michelson <mmichelson@digium.com> - - * main/utils.c: There existed about a 1 in 4 billion chance that - reading from /dev/urandom would return LONG_MIN (1 in 9 - quintillion if using 64-bit longs). Since there is no positive - equivalent of LONG_MIN, the result of labs() in this case is - unpredictable. This fixes that situation. (closes issue #11336, - reported and patched by sperreault) - -2007-11-21 16:24 +0000 [r89484] Russell Bryant <russell@digium.com> - - * channels/chan_unistim.c: Fix some code that was supposed to - ensure that a buffer was terminated, but was writing to the wrong - byte. Also, remove some non-thread safe test code. (closes issue - #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by - IgorG (license 20) - additional changes by me - -2007-11-21 16:08 +0000 [r89483] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I - attempted to port to trunk as well. Unfortunately, since trunk - uses read/write locks for the context lock, it means that I have - actually *introduced* a deadlock condition since they are not - recursive. Removing this change for now and will look into - introducing a different one. - -2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c, - utils/conf2ael.c: move these forward declarations back to - asterisk.h where they belong... even though asterisk.h includes - compat.h, these declarations have nothing to do with the being - platform-compatible and are directly related to being part of - Asterisk - - * channels/chan_usbradio.c: get this to actually compile... - - * main/pbx.c: remove some debugging code that doesn't compile on - 64-bit platforms - -2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy <murf@digium.com> - - * res/res_features.c: OOOps! All the debug stuff I inserted was - accidentally committed. I hereby revert it. - - * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks - to snuffy for his work on neatening up the code and removing - duplicated code. - -2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo <rizzo@icir.org> - - * channels/gentone-ulaw.c (removed): remove this file, it is not - used anywhere. - - * main/astmm.c: add missing paths.h - - * configure, include/asterisk/autoconfig.h.in, configure.ac: add - check for video4linux - -2007-11-21 01:09 +0000 [r89474] Steve Murphy <murf@digium.com> - - * main/pbx.c: A free in add_pri was ultimately the source of the - grief we were having with parking. This set of changes fixes that - problem, and introduces some more error messages, and puts debugs - into ifdefs for what could be short-term usage. Txs to Terry W. - for his help, guidance, and especially patience. - -2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo <rizzo@icir.org> - - * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c, - main/minimime/mm.h, utils/check_expr.c: more header - removal/normalization - - * configure, include/asterisk/autoconfig.h.in, configure.ac: X11 - checks (at least some - for other platforms with unusual X11 - locations you might need to add more directories) - -2007-11-21 00:21 +0000 [r89470] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c, CHANGES: Merge changes from - team/russell/sla_trunk_moh ... * Added the ability to specify the - music on hold class used to play into the conference when there - is only one member and the M option is used. * Added the ability - to specify a music on hold class to play instead of ringing for - the SLATrunk application. (patched by me, and tested internally) - -2007-11-21 00:20 +0000 [r89469] Luigi Rizzo <rizzo@icir.org> - - * makeopts.in: complete support for X11 - -2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk - build again - - * main/say.c: Add support for new recorded character sounds Closes - issue #5208 - -2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, - apps/app_dictate.c, apps/app_test.c, apps/app_ices.c, - apps/app_followme.c, channels/chan_iax2.c, main/config.c, - main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c, - main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h, - res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c, - main/app.c, main/image.c, apps/app_directory.c, main/db.c, - cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c, - include/asterisk/utils.h, include/asterisk.h, main/pbx.c, - channels/chan_sip.c, res/res_crypto.c, - include/asterisk/channel.h, res/res_monitor.c, - include/asterisk/paths.h, main/file.c, apps/app_sms.c, - include/asterisk/ael_structs.h, pbx/pbx_config.c, - apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c: - move asterisk/paths.h outside asterisk.h and into those files who - really need it. - - * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c, - main/threadstorage.c, main/devicestate.c, - include/asterisk/_private.h (added), main/astobj2.c, - main/loader.c, main/term.c, main/cli.c, main/channel.c, - main/manager.c, main/logger.c, build_tools/strip_nonapi, - main/event.c, main/asterisk.c, main/db.c: move internal function - declarations to include/asterisk/_private.h - -2007-11-20 19:29 +0000 [r89464] Russell Bryant <russell@digium.com> - - * configure, configure.ac: i got a little carried away with commas - ... - -2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/module.h, build_tools/make_buildopts_h, - main/loader.c: switch compile-time option checking to string - storage mode in this branch too - -2007-11-20 19:11 +0000 [r89460] Russell Bryant <russell@digium.com> - - * configure, configure.ac: fix the zaptel configure script check - -2007-11-20 18:20 +0000 [r89459] Luigi Rizzo <rizzo@icir.org> - - * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before - regenerating configure, i have more changes) - -2007-11-20 17:59 +0000 [r89458] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, /: Merged revisions 89457 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov - 2007) | 9 lines According to comments in main/pbx.c, it is - essential that if we are going to lock the conlock as well as the - hints lock, it must be locked in that respective order. In order - to prevent a potential deadlock, we need to lock the conlock - prior to locking the hints lock in ast_hint_state_changed (see - the call stack example on issue #11323 for how this can happen). - (closes issue #11323, reported by eelcob, suggestion for patch by - eelcob, patch by me) ........ - -2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo <rizzo@icir.org> - - * makeopts.in: prepare to support console_video - - * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile, - channels/Makefile: Fix building of modules under cygwin. After - this commit we can actually load modules under windows, and we - can start debugging more interesting problems related to the load - order and functionality of modules. - -2007-11-20 16:11 +0000 [r89453] Mark Michelson <mmichelson@digium.com> - - * configs/sip.conf.sample: Changed occurrences of "busy-level" to - "busylevel" in sip.conf.sample in light of commit 89441. Thanks - to pj for pointing out the need for this (closes issue #11307, - reported by pj) - -2007-11-20 15:39 +0000 [r89452] Luigi Rizzo <rizzo@icir.org> - - * configure, configure.ac, acinclude.m4: add an argument for extra - headers to AC_EXT_LIB_CHECK, and on passing simplify the code. - Too bad that every time we need to regenerate configure... - -2007-11-20 15:30 +0000 [r89451] Steve Murphy <murf@digium.com> - - * /, doc/tex/queues-with-callback-members.tex: Merged revisions - 89450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 - line closes issue #11324; break statements missing in switch - cases. ........ - -2007-11-20 15:00 +0000 [r89449] Joshua Colp <jcolp@digium.com> - - * main/translate.c: Minor documentation tweak and if an incorrect - parameter is given to core show translation return the usage - information. (closes issue #11316) Reported by: eliel Patches: - translate.c.patch uploaded by eliel (license 64) - -2007-11-20 14:54 +0000 [r89448] Luigi Rizzo <rizzo@icir.org> - - * configure, acinclude.m4: comment a bit the code in acinclude.m4 - There is still a lot of code to clean up there, but hopefully - this should clarify what goes on in there. - -2007-11-20 14:49 +0000 [r89447] Joshua Colp <jcolp@digium.com> - - * channels/h323/ast_h323.cxx: Include the compatibility header file - in ast_h323.cxx for compatibility reasons. (closes issue #11311) - Reported by: falves11 - -2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fix sip show history. Closes issue #11312 - - * channels/chan_sip.c: Change terminology a bit for CLI commands - handling SIP channels/calls/dialogs/whatever. Closes issue #11312 - -2007-11-20 07:42 +0000 [r89443] Luigi Rizzo <rizzo@icir.org> - - * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile - changes to build loadable modules under cygwin (not complete yet - - still need to sort out dependecies on res_*) - -2007-11-20 00:17 +0000 [r89442] Steve Murphy <murf@digium.com> - - * main/pbx.c: Get rid of some debug messages in pbx.c - -2007-11-19 23:24 +0000 [r89441] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in - sip.conf to "busylevel" to be more parallel with the SIPPEER() - argument of the same name. The deprecation procedure is not being - used here since this is a trunk-only option. (closes issue - #11307, reported by pj, patched by me) - -2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant <russell@digium.com> - - * include/asterisk/module.h: Be a bit more pedantic about the type - for holding the md5 sum for the build options. Also, doxygenify - the comment. - - * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect - which options were compiled in - -2007-11-19 22:55 +0000 [r89438] Steve Murphy <murf@digium.com> - - * main/pbx.c: These changes were made in response to - niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in - two different contexts, included in the same context. In that - particular case, we were behaving the same as 1.4, but after - experimenting, I quickly found that if 20 and 201 were in the - same extension, 1.4 would return 201, and this code returns 20. - These changes now enable the current code to replicate the - behavior of 1.4 in respect to MATCHMORE in cases like this. - -2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c: - another few errno.h removals - - * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, - apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c, - pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c, - apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c, - apps/app_milliwatt.c: more errno.h removal - - * funcs/func_sysinfo.c: remove unnecessary headers - - * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary - includes. - -2007-11-19 20:13 +0000 [r89429] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Change delimiter of SIPPEER to be comma - (instead of pipe) and further deprecate the old ':' delimiter - Reported by: pj Patch by: tilghman Closes issue #11305 - -2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo <rizzo@icir.org> - - * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c, - codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, - codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, - codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless - includes from codecs - - * formats/format_ilbc.c, formats/format_sln.c, - formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, - formats/format_ogg_vorbis.c, formats/format_g723.c, - formats/format_h263.c, formats/format_h264.c, - formats/format_wav_gsm.c, formats/format_g726.c, - formats/format_jpeg.c, formats/format_gsm.c, - formats/format_g729.c: format handlers don't need network, lock, - channel and scheduler headers - - * include/asterisk.h, include/asterisk/compat.h, - include/asterisk/lock.h, utils/extconf.c, - include/asterisk/abstract_jb.h: move the declaration of struct - ast_channel ast_frame and ast_module to compat.h so it is always - available - hopefully this will let us reduce the number of - inclusions of channel.h and frame.h - - * main/udptl.c, main/autoservice.c, funcs/func_rand.c, - cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c, - main/threadstorage.c, main/say.c, funcs/func_env.c, - funcs/func_strings.c, main/devicestate.c, - cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c, - main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c, - main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c, - main/sched.c, main/pbx.c, funcs/func_timeout.c, - funcs/func_math.c, funcs/func_cut.c, main/chanvars.c, - main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c, - funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c, - funcs/func_iconv.c, main/alaw.c, main/asterisk.c, - funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c, - main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c, - funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c, - funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c, - main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c, - funcs/func_groupcount.c, main/manager.c, main/tdd.c, - funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c, - main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c, - funcs/func_db.c, main/dnsmgr.c, main/translate.c, - main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c, - cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c, - funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c, - main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c, - funcs/func_callerid.c, main/dsp.c: another bunch of include - removals (errno.h and asterisk/logger.h) - - * channels/chan_local.c, apps/app_record.c, - apps/app_alarmreceiver.c, apps/app_chanisavail.c, - apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, - channels/chan_skinny.c, formats/format_pcm.c, - apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c, - codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c, - codecs/codec_g726.c, formats/format_ogg_vorbis.c, - apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c, - apps/app_speech_utils.c, apps/app_sendtext.c, - formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c, - apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c, - res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c, - apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c, - channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c, - res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c, - apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c, - apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c, - apps/app_adsiprog.c, codecs/codec_lpc10.c, - res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c, - codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c, - channels/chan_jingle.c, apps/app_channelredirect.c, - apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c, - formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c, - apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c, - apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c, - formats/format_wav_gsm.c, res/res_smdi.c, - include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c, - channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c, - formats/format_jpeg.c, formats/format_gsm.c, - apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, - apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, - channels/chan_agent.c, apps/app_disa.c, - channels/iax2-provision.c, res/res_ael_share.c, - apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c, - channels/chan_misdn.c, apps/app_waitforring.c, - apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c, - apps/app_zapateller.c, res/res_indications.c, - codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c, - apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, - codecs/codec_zap.c, res/res_config_sqlite.c, - channels/misdn_config.c, apps/app_controlplayback.c, - formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c, - main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, - res/res_clioriginate.c, apps/app_while.c, include/asterisk.h, - apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c, - res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, - res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c, - apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c, - apps/app_flash.c, apps/app_directed_pickup.c, - channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c, - apps/app_senddtmf.c, formats/format_g729.c: include "logger.h" - and errno.h from asterisk.h - usage shows that they were included - almost everywhere. Remove some of the instances. - -2007-11-19 17:18 +0000 [r89422] Steve Murphy <murf@digium.com> - - * main/pbx.c: a correction to code involved in an extension removal - -2007-11-19 16:29 +0000 [r89421] Mark Michelson <mmichelson@digium.com> - - * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan - function for retrieval of system information - -2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 89419 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6 - lines Print out the correct filename (features.conf) in the log - message when parkpos options are incorrect. (closes issue #11295) - Reported by: Laureano Patches: res_features.c.patch uploaded by - Laureano (license 265) ........ - - * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov - 2007) | 4 lines Clarify documentation a bit, include that a frame - has to pass through the core in order for the Local channel - optimization to happen. (closes issue #11246) Reported by: jon - ........ - -2007-11-19 14:36 +0000 [r89412] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/logger.h: revert inclusion of options.h - -2007-11-19 14:03 +0000 [r89410] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c: Change warning messages (which are really - debug messages) into debug messages. (closes issue #11288) - Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded - by IgorG (license 20) - -2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson <oej@edvina.net> - - * CHANGES: Update CHANGES - - * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan - function. With this, you can control the peer in the dialplan, so - you avoid placing outbound calls when the device has reached - busy-level. Reported by pj. Closes bug #11180 - - * main/acl.c: Add some debugging to the routines that finds our - local IP address. Related to bug #9225 - - * channels/chan_sip.c: Make some notes about a problem I found with - the OPTIONs handler while working with the bug tracker. Since we - don't authenticate devices (peers/users) on OPTIONS we don't have - the proper context set for the user/peer. However, we might not - want to process an authentication for every OPTIONS, so we could - have a config option for this, "optionsforceok" to always answer - 200 OK on the request and not check device or destination, nor - add a SDP. If Asterisk sends the OPTIONs request, it doesn't care - about the reply. Some devices use OPTIONs to discover - capabilities, since we should answer like an INVITE from the - device and we need to support that properly too, which we don't - today. So much to do :-) - -2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp <jcolp@digium.com> - - * build_tools/make_buildopts_h: Add OSX into the logic that uses - md5 instead of md5sum. - - * include/asterisk/compat.h: Use the easy way that rizzo mentioned, - only include malloc.h on the Windows platform. - - * include/asterisk/compat.h: Revert last commit, apparently - buildbot lied to me. - - * include/asterisk/compat.h: Change how we handle alloca to conform - with how it is suggested in the autoconf manual for - AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms - should be broken by this. - - * configure, configure.ac: Change autoconf logic a bit so it says - what it is looking for in two instances where it didn't. - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/lock.h, include/asterisk/network.h: Use autoconf - logic to determine the presence of - PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and - PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h - in " - -2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add SS7 Generic address support (#11156) - -2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/compat.h: if alloca.h is not present, try - malloc.h - - * agi/Makefile: temporarily disable this target in mingw - - * Makefile: will i ever get precedences for windows right ? in the - meantime, use a variable to ease enabling/disabling print - subdirectories. - - * Makefile: reformulate dependencies in a more correct way - -2007-11-17 17:46 +0000 [r89388] Steve Murphy <murf@digium.com> - - * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make - it so it will compile. Hope I did the right thing. And some - additions to removal of extens to take care of hashtab pointers - in all cases. - -2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo <rizzo@icir.org> - - * Makefile.moddir_rules, Makefile.rules: as discussed some time ago - on the -dev list, create embedde object with a .eo suffix even if - they are coming from .cc sources. This simplifies the handling in - the build scripts. - - * include/asterisk/network.h: prefer socket.h over other variants - (winsock etc.) - - * channels/chan_local.c, main/translate.c, - channels/chan_features.c, main/http.c, main/config.c: trim more - redundant headers - - * main/acl.c: remove unnecessary includes - - * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c, - main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by - previous mistake - - * Makefile: wrong variable, wrong order -> broken build. - - * include/asterisk/acl.h, include/asterisk/utils.h, - include/asterisk/autoconfig.h.in, include/asterisk/rtp.h, - configure.ac, main/acl.c, include/asterisk/netsock.h, - main/utils.c, include/asterisk/manager.h, main/netsock.c, - main/manager.c, res/res_agi.c, pbx/pbx_dundi.c, - include/asterisk/udptl.h, include/asterisk/dnsmgr.h, - main/asterisk.c: start using asterisk/network.h for network - related headers. Also remove some unnecessary includes. - - * include/asterisk/network.h (added): wrapper for all generic - network headers that have different names and locations on the - various systems. - - * main/cygload.c: main is called main not amain! - - * main/Makefile: conditional targets for building the windows - version - - * Makefile: support cygwin targets - - * Makefile.moddir_rules: and this is the last one to have asterisk - compile (not run yet) natively under cygwin. - - * apps/app_sms.c: another cygwin compatibility fix. This one must - be handled in a better way in configure, also for other - architectures - - * utils/Makefile, main/Makefile, utils/extconf.c: more - cygwin/mingw32 compatibility fixes - - * include/asterisk/channel.h: use autoconf results to conditionally - compile timersub - - * include/asterisk/lock.h: compatibility fixes for cygwin - - * include/asterisk/compat.h: some version of flex produce code that - wants __STDC_VERSION__ defined, but the compiler does not always - define it. - - * Makefile: these linker flags apply to both cygwin and mingw32 - - * utils/hashtest2.c: add a return NULL to a function that is - expected to return a value so compilers that don't understand - that this code is NOTREACHED will not complain (the fault is not - much on the compiler but on the declaration of pthread_exit on - certain platforms) s/certain platform/cygwin/ if you are really - curious - - * main/loader.c: define RTLD_LOCAL for platforms that don't have - it. This is only to complete the build, clearly the linker - behaviour will be completely different and likely to cause - trouble in those cases. - - * channels/Makefile: filter out modules that do not compile under - windows (this should be handled with the dependencies generated - by configure and menuselect, but will be fixed later) - - * main/utils.c: netdb.h is used for gethostbyname, and it was not - included in some platforms. - - * main/cygload.c (added): Loader for cygwin where asterisk is - really a big dll (something like this is already in 1.2) - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - timersub is a macro not a function, so write the check in a way - that detects both formats. - -2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant <russell@digium.com> - - * pbx/pbx_lua.c: fix the build of pbx_lua - - * configure, include/asterisk/autoconfig.h.in, - include/asterisk/compat.h, configure.ac, include/asterisk/io.h, - include/asterisk/channel.h: Update the configure script check for - sys/poll.h to also provide the result in - include/asterisk/autoconfig.h. Also, move the conditional include - of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h - instead of the two headers it existed in before. - - * build_tools/make_buildopts_h: actually let this compile, oops :( - - * build_tools/make_buildopts_h: Use the fix suggested by Tilghman - on the -dev to make cutting up the BUILDSUM friendly to non-bash - shells. I think this should work for BSD/mingw as well, but did - not yet remove the switch statement. - -2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo <rizzo@icir.org> - - * Makefile: linker flags for mingw32 - - * configure, include/asterisk/autoconfig.h.in, configure.ac: add - detection for timersub() and winsock.h/winsock2.h - - * include/asterisk/endian.h: provide definitions for - __LITTLE_ENDIAN and __BIG_ENDIAN if not present. - - * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h: - use poll as detected by configure - - * configure, configure.ac, makeopts.in: use autoconf to check for - the existence of sys/poll.h - - * build_tools/make_buildopts_h: this script is run on the build - system, not on the host. - - * Makefile.moddir_rules: compatibility fix for mingw32 - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4, makeopts.in: acinclude.m4: add a function to help - checking sdl-config, gtk-config and the like (this could be used - for gtk and gtk2 as well) Other files: add tests for sdl, - sdl_image and avcodec and regenerate configure and - autoconfig.h.in - - * include/asterisk/autoconfig.h.in, configure.ac: add check for the - presence of glob - - * channels/chan_jingle.c, channels/chan_unistim.c, - funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c, - channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c, - utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, - apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate - #include lines - - * main/udptl.c, funcs/func_module.c, res/res_features.c, - funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c, - channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c, - pbx/pbx_loopback.c, funcs/func_blacklist.c, - channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c, - main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, - apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c: - remove a bunch of duplicate includes Reproduce with grep -r - #include . | grep -v .svn | grep -v Binary | sort | uniq -c | - sort -nr - -2007-11-16 23:44 +0000 [r89347] Terry Wilson <twilson@digium.com> - - * res/res_features.c: Fix broken parking dial-back - -2007-11-16 23:33 +0000 [r89346] Steve Murphy <murf@digium.com> - - * main/pbx.c: My goodness, haven't handled an extension deletion. - Add code to ast_context_remove_extension2() to remove an - extension from the trie. Done by marking it deleted. The - scoreboard won't update for it any more. Also, a couple of calls - to insert hashtab had a spurious ->exten, which was removed. - -2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/paths.h, include/asterisk.h: paths are already - in include/asterisk/paths.h so don't duplicate them in - include/asterisk.h - - * include/asterisk/utils.h, include/asterisk/lock.h: whitespace - only change - adjust indentation and add some comments on the - content of these two files. utils.h (which is included in over - 150 files) contains a lot of unrelated functions which require - the inclusion of a large number of other headers. At some point - we should partition its content in a better way. - -2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/logger.h: logger.h does not need options.h - - * include/asterisk/utils.h, channels/chan_sip.c, - include/asterisk/astobj.h, include/asterisk/compat.h, - include/asterisk/channel.h, include/asterisk/strings.h, - utils/extconf.c, include/asterisk/frame.h, - include/asterisk/stringfields.h, include/asterisk/endian.h: - remove redundant #include "asterisk/compat.h", but make sure that - asterisk/compiler.h is included everywhere - - * main/acl.c, main/asterisk.c: remove duplicate headers. Properly - check for netdb.h (there is actually tens of places to fix) - - * Makefile.rules: put back default optimization to -O6 (previously - changed by mistake) - - * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c, - apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, - channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c, - apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c, - include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c, - main/pbx.c, funcs/func_timeout.c, formats/format_wav.c, - apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c, - apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c, - main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c, - main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h, - apps/app_voicemail.c, formats/format_sln.c, - apps/app_authenticate.c, apps/app_readexten.c, - apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules, - apps/app_setcallerid.c, include/asterisk/astmm.h, - res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c, - apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c, - main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c, - funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, - res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c, - apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c, - main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c, - apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c, - main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c, - cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c, - formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c, - funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c, - main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c, - formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, - main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c, - channels/iax2-provision.c, res/res_monitor.c, main/netsock.c, - apps/app_waitforring.c, main/fixedjitterbuf.c, - include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c, - channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c, - apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c, - res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, - res/res_config_sqlite.c, main/fskmodem.c, - channels/misdn_config.c, apps/app_controlplayback.c, - cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c, - channels/chan_phone.c, funcs/func_enum.c, main/dial.c, - main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c, - main/logger.c, main/image.c, apps/app_ivrdemo.c, - res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c, - channels/chan_zap.c, main/slinfactory.c, res/res_convert.c, - pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c, - channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, - channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c, - include/asterisk/strings.h, apps/app_senddtmf.c, - funcs/func_callerid.c, include/asterisk/time.h, - channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c, - apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c, - pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c, - main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c, - apps/app_morsecode.c, formats/format_ogg_vorbis.c, - channels/iax2-parser.c, apps/app_speech_utils.c, - include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c, - funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c, - utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c, - formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c, - channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, - res/res_jabber.c, funcs/func_uri.c, main/io.c, - include/asterisk/abstract_jb.h, main/channel.c, - apps/app_minivm.c, res/res_realtime.c, main/dns.c, - apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, - codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c, - res/res_adsi.c, include/asterisk/plc.h, - apps/app_channelredirect.c, formats/format_vox.c, - main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c, - funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c, - cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c, - apps/app_sayunixtime.c, apps/app_test.c, main/config.c, - main/loader.c, main/term.c, main/cli.c, res/res_smdi.c, - include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c, - pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, - include/asterisk/utils.h, apps/app_privacy.c, - codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, - funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c, - apps/app_transfer.c, include/asterisk/manager.h, - apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c, - channels/chan_misdn.c, funcs/func_curl.c, - channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c, - res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c, - include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c, - funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c, - include/asterisk/file.h, main/tdd.c, funcs/func_volume.c, - res/res_agi.c, main/app.c, apps/app_parkandannounce.c, - cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c, - res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, - main/translate.c, include/asterisk/config.h, main/jitterbuf.c, - main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c, - funcs/func_extstate.c, apps/app_directed_pickup.c, - main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c, - main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c, - formats/format_g729.c: Start untangling header inclusion in a way - that does not affect build times - tested, there is no - measureable difference before and after this commit. In this - change: use asterisk/compat.h to include a small set of system - headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, - stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the - inclusion is conditional on HAVE_FOO_H as determined by autoconf. - Normally, source files should not include any of the above system - headers, and instead use either "asterisk.h" or - "asterisk/compat.h" which does it better. For the time being I - have left alone second-level directories (main/db1-ast, etc.). - -2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson <mmichelson@digium.com> - - * main/manager.c: Fixing a problem pointed out by Qwell - - * main/manager.c: Added some locks that should have been around - astman_send_error, at least according to the comments. (closes - issue #11258, reported and patched by eliel) - -2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy <murf@digium.com> - - * main/pbx.c: This corrects a hashtab removal, given a bad argument - - * main/pbx.c, res/res_features.c: This fixes a problem with pattern - ranges; and corrects a situation in res_features, where an - extension would be created with the name Zap/51, as an example. - THe / is bad because it would tend to mean that the 51 is to be - cid matched. - -2007-11-16 18:48 +0000 [r89328] Luigi Rizzo <rizzo@icir.org> - - * build_tools/make_buildopts_h: both md5sum and variable - substitutions such as ${BUILDSUM:0:8} are not available in - FreeBSD. For the time being, put in a workaround so we can build - the system, and wait for the result of the discussion on whether - we can store the md5 as a string rather than 4 ints (if so, we - won't need more complex tricks with awk or sed for splitting the - md5). 1.4 will be fixed when we decide the issue. - -2007-11-16 17:11 +0000 [r89327] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Adding confirmation playback when - forwarding voicemail messages. This will attempt to play the - name(s) of the person(s) to whom you are forwarding the message - prior to prompting for prepending. If no name is found, the - extension is read back verbatim. (closes issue #9046, reported - and patched by jaroth) - -2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/module.h, build_tools/make_buildopts_h, - main/loader.c: Merged revisions 89325 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) - | 4 lines To help combat problems where people build external - modules (asterisk-addons or others) and then change the build - options of the Asterisk build in a way that makes the - incompatible without warning, this commit introduces an MD5 - signature of the important build-time options and includes that - signature into modules when they are built. When the loader loads - one of these modules and notices the problem, it will emit a - warning to console and refuse to initialize the module, as doing - so could cause the system to be unstable or even crash. If you - upgrade to this version of Asterisk, you must rebuild *all* of - your modules that came from other sources before trying to run - this version. If you are using Digium's G.729 binary codec - module, you will need v33 or newer. ........ - -2007-11-16 15:44 +0000 [r89324] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov - 2007) | 5 lines Make realtime queues accessible from the - QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and - patched by atis, with small modifications from me) ........ - -2007-11-16 10:07 +0000 [r89322] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/config.h, main/config.c: add a small new - function to retrieve variables from a config once we have a - pointer to the category. - -2007-11-16 10:06 +0000 [r89321] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed #10631, about one way audio. thanks - IgorG again. - -2007-11-16 09:51 +0000 [r89320] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: move the inner part of config file parsing - to a separate function, so it can be reused in the implementation - of cli commands when they have a similar syntax. - -2007-11-16 08:54 +0000 [r89319] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269, - thanks IgorG. - -2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher <tlesher@digium.com> - - * main/utils.c, include/asterisk/stringfields.h: If we're going to - be passing a negative value for the size of a stringfield, in - order to indicate something, then using an UNSIGNED parameter is - bad, mmmmmkay? - - * Makefile, /: Merged revisions 89302 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007) - | 2 lines Start Asterisk in Debian at a more reasonable time - (since zaptel is at level 20) ........ - - * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 - Nov 2007) | 2 lines Fix an uninitialized memory read found by - valgrind ........ - - * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being - renamed. - - * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) - | 5 lines Yet another memory corruption issue. Reported by: atis - Patch by: tilghman Fixes issue #10923 ........ - -2007-11-15 17:27 +0000 [r89297] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | - 8 lines Update the SLAStation application to account for the case - where the SLA thread has a call out to the station, but the user - has pressed a line button to answer the call instead of picking - up the handset. If they do, the phone sends out a new INVITE. So, - the SLAStation app must check to see if it is picking up a - ringing trunk, and ensure that the other stations stop ringing. - (reported internally, patched by me, tested by mogorman) ........ - -2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy <murf@digium.com> - - * main/pbx.c: Get rid of a previously missed ast_log call for - debug, no longer nec. - - * main/pbx.c: Perhaps I went overboard on initializing things. I - can remove unnecc. stuff later. A few bug fixes. Killing small - bugs on the way to killing bigger ones. Removed locking on - hashtabs; there's plenty of locks already being taken. A small - bug in the root_tree hashtab compare func. - -2007-11-15 16:20 +0000 [r89293] Luigi Rizzo <rizzo@icir.org> - - * main/channel.c, apps/app_channelredirect.c, main/manager.c, - res/res_features.c, apps/app_softhangup.c, - include/asterisk/channel.h, include/asterisk/lock.h, - apps/app_senddtmf.c: access channel locks through - ast_channel_lock/unlock/trylock and not through ast_mutex - primitives. To detect all occurrences, I have renamed the lock - field in struct ast_channel so it is clear that it shouldn't be - used directly. There are some uses in res/res_features.c (see - details of the diff) that are error prone as they try and lock - two channels without caring about the order (or without - explaining why it is safe). - -2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp <jcolp@digium.com> - - * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been - used, not 'increasing'. - - * channels/chan_sip.c, channels/chan_h323.c, - channels/misdn_config.c: And file said... let trunk build again! - Accomplished by some more constification, and marking a function - in chan_sip as purposely unused until it is fixed up. - -2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, /: Merged revisions 89288 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov - 2007) | 3 lines Undoing previous commit since I realize it was - wrong ........ - - * main/manager.c, /: Merged revisions 89286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov - 2007) | 4 lines Adding a missing mutex unlock. (closes issue - 11256, reported and patched by ys) ........ - -2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Always relying on the responses when - crossing NAT's are not a good solution, it breaks communication. - Rizzo - you need to implement a configuration option for this - code. It's good, but maybe should be off by default. - - * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 - lines Don't send re-invites during pending INVITE transactions. - Patch by one47 - thanks! Closes issue #9305 ........ - - * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 - lines Improve support for multipart messages. Code by gasparz, - changes by me (mostly formatting). Thanks, gasparz! Closes issue - #10947 ........ - - * channels/chan_sip.c: Exit early instead of deciding to exit after - processing the message. - - * channels/chan_sip.c, configs/sip.conf.sample: Add support for - application/dtmf SIP INFO dtmf handling. Yep, another way of - handling DTMF in SIP. Totally undocumented, but implemented in - enough devices so we have to support it. Code by sergee, small - changes by oej. Closes issue #11049 - -2007-11-15 01:42 +0000 [r89277] Steve Murphy <murf@digium.com> - - * main/pbx.c: Had trouble playing with parking; spent a long time - trying to reason out MATCHMORE mode. made these updates and xfers - on zaptel lines seem to work ok now - -2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: Merged revisions 89275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007) - | 5 lines When a recording ends with '#', we are improperly - trimming an extra 200ms from the recording. Reported by: sim - Patch by: tilghman Closes issue #11247 ........ - - * main/channel.c: Typo - - * main/channel.c: Add callerid to the Hangup manager event. - Reported by: outtolunc Patch by: outtolunc Closes issue #11248 - -2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy <murf@digium.com> - - * main/pbx.c: Rescaled the weights of the patterns to give - something more independent of pattern length; and make . less - likely to win. Question: which should win for 14102241145-- - _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will - win. - - * main/pbx.c: A further problem highlighted by 11233 has been - resolved; a certain combination of patterns in a certain order, - led to a malformed trie, due to a ptr not being initialized in - the loop. Also, some tree printing prettifications. - -2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c, - res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more - typo in config.c; and missed conversions due to the constifying - of ast_variable_new parameters - - * main/config.c: Typo - -2007-11-14 13:18 +0000 [r89268] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/acl.h, channels/chan_sip.c, - include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c, - main/acl.c, pbx/dundi-parser.c, apps/app_queue.c, - channels/chan_iax2.c, main/enum.c, channels/chan_oss.c, - apps/app_playback.c, main/config.c, pbx/dundi-parser.h, - include/asterisk/abstract_jb.h, main/manager.c, - channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c, - main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c, - apps/app_voicemail.c: make the 'name' and 'value' fields in - ast_variable const char * This prevents modifying the strings in - the stored variables, and catched a few instances where this was - actually done. Given the differences between trunk and 1.4 (and - the fact that this is effectively an API change) it is better to - fix 1.4 independently. These are chan_sip.c::sip_register() - chan_skinny.c:: near line 2847 config.c:: near line 1774 - logger.c::make_components() res_adsi.c:: near line 1049 I may - have missed some instances for modules that do not build here. - -2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant <russell@digium.com> - - * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding - guidelines issues ... - handle memory allocation failures - add - an ast_ prefix to a publicly exported function - put curly braces - in the right places - add a bunch of spaces where they should be - be used - - * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple - places - return errors from load_module / unload_module - - * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the - syntax highlighting in my editor happy - - * apps/app_queue.c: Instead of reserving 800 bytes for periodic - announcements, use an array of ast_str pointers and only alloate - space for the strings as needed. - -2007-11-14 01:16 +0000 [r89262] Joshua Colp <jcolp@digium.com> - - * main/srv.c, /: Merged revisions 89260 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4 - lines Return the proper value when the srv_callback function - executes properly. (closes issue #11240) Reported by: jtodd - ........ - -2007-11-14 01:15 +0000 [r89261] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Convert most of the strings in the call_queue - struct to use stringfields. - -2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, main/pbx.c: use simpler technique for removing - known entries from lists - -2007-11-14 00:33 +0000 [r89258] Russell Bryant <russell@digium.com> - - * main/image.c: - Simplify removing an item from a list - move a - verbose message to after the item is added to the list - make use - of the ARRAY_LEN macro in one spot - -2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy <murf@digium.com> - - * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't - covered the case of a context with no patterns. (blush) - - * main/pbx.c: closes issue #11233 -- where some fine points in the - algorithm to build the tree needed to be corrected. Many thanks - for the test case, jtodd - -2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: This fixes a build error on my mac. It - also works on my linux box. Let me know if it breaks any other - platform ... - - * res/res_features.c: Fix a typo pointed out by outtolunc, thanks - :) - - * channels/chan_sip.c: - Convert initialization of a struct to C99 - style instead of GNU style - Fix a minor spelling error in a - comment - - * res/res_features.c, CHANGES: Update the ParkedCall application to - grab the first available parked call if no parked extension is - provided as an argument. (closes issue #10803) Reported by: - outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by - outtolunc (license 237) - modified by me to work a bit - differently ... - -2007-11-13 19:48 +0000 [r89249] Jason Parker <jparker@digium.com> - - * /, res/res_features.c: Merged revisions 89248 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600 - (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064. - It is documented behavior that if a parking extension already - exists while using PARKINGEXTEN, dialplan execution will - continue. If blind transferring to a Park with PARKINGEXTEN, you - must keep this in mind, and handle the failure yourself. Issue - 11237, reported by jon. ........ - -2007-11-13 17:41 +0000 [r89247] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) - | 2 lines If we set a value for qualify, we should actually pay - attention to it, instead of overriding the value ........ - -2007-11-13 16:03 +0000 [r89242] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 - Nov 2007) | 5 lines Reverting commit made in revision 89205 since - it is unnecessary. Thanks to Kevin for pointing this out ........ - -2007-11-13 14:03 +0000 [r89240] Tilghman Lesher <tlesher@digium.com> - - * /, main/utils.c: Merged revisions 89239 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007) - | 4 lines Debugging is running into the 16-lock limit. Increase - to avoid. (This define is only effective when debugging is turned - on, so there's no effect for most installations.) ........ - -2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson <mmichelson@digium.com> - - * apps/app_mixmonitor.c: There is the potential to copy - uninitialized memory into the mixmonitor->post_process string. - This fix prevents that. - - * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 - Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only - 1 argument is given, then the args.options and args.post_process - strings are uninitialized and could contain garbage. This change - handles this situation properly by only using arguments that we - have parsed. ........ - -2007-11-13 00:19 +0000 [r89202-89203] Jason Parker <jparker@digium.com> - - * Makefile: oops, somebody left out the directory here... - - * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c, - include/asterisk/config.h, res/res_convert.c, res/res_crypto.c, - pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h, - res/res_config_sqlite.c, res/res_monitor.c, - include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen - fixes. Also fix a common typo I kept seeing (arguement) in - various files. Closes issue #11222, patch by snuffy (with - arguement > argument by me). - -2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy <murf@digium.com> - - * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake - of the mtx_prof stuff. - - * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen - update to hashtab.h; closes issue #11223 - - * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work, - who brought up that these fixes might need to be made. - -2007-11-12 20:48 +0000 [r89195] Jason Parker <jparker@digium.com> - - * main/pbx.c, /: Merged revisions 89194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1 - line Fix a typo pointed out by De_Mon on #asterisk-dev ........ - -2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming <kpfleming@digium.com> - - * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported - by: eliel Patches: utils.Makefile.patch uploaded by eliel - (modified by me) (license 64) - -2007-11-12 18:44 +0000 [r89186] Steve Murphy <murf@digium.com> - - * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, - funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, - apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in - asterisk-dev by Brian Capouch, I determined I too agressive in - not initializing arrays passed to pbx_substitute_variables_xxxx; - I reviewed the code (again) and hopefully found every possible - spot where substitute_variables is called conditionally, and made - sure the char array involved was set to a null string. - -2007-11-12 17:44 +0000 [r89185] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) - | 5 lines Fix two cases of memory corruption caused by background - threads. Reported by: atis Patch by: tilghman Fixes issue #10923 - ........ - -2007-11-12 13:36 +0000 [r89178-89179] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged - revisions 89173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | - 1 line if we're NT and no number was dialed and overlapdial is - set, we wait for the ISDN timeout instead of starting our own - timer. added a comment for the misdn.conf.sample for the - overlapdial config option. ........ - - * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - Merged revisions 89172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | - 1 line added restart all interfaces Restart_Indicator, to - automatically send a RESTART after the L2 of a PTP Port comes up. - Also fixed some places where we have send a RELEASE without need - for it. ........ - -2007-11-12 13:26 +0000 [r89177] Joshua Colp <jcolp@digium.com> - - * channels/chan_unistim.c, utils/hashtest.c: Fix building on - FreeBSD by including/not including some headers. (closes issue - #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys - (license 281) - -2007-11-12 13:22 +0000 [r89174-89176] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 89171 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | - 1 line fixed a state/event issue with overlapdial=yes when no - extension matched. removed the general sending of a - RELEASE_COMPLETE when we receive a RELEASE, this is done by - mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with - mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to - upgrade to at least mISDNuser-1.1.6 (when using the NT mode at - all) ........ - - * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 - Nov 2007) | 1 line fixed the support for CW and therefore for the - reject_cause option. ........ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, - channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged - revisions 89169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | - 1 line aded ntkeepcalls option, to avoid droÃpping calls when the - L2 goes down on a PTP link. There are some pbx which do turn off - the L1 for a very short while and restart it immediately. - normally T310 should be started and after 10 seconds or so the - calls should be dropped, this is a simple fix wihtout this timer. - ........ - -2007-11-09 18:57 +0000 [r89130-89132] Jason Parker <jparker@digium.com> - - * configs/usbradio.conf.sample (added): Add usbradio.conf.sample - from branches/1.4/configs - r84162. It was mistakenly deleted in - 1.4 without ever being merged to trunk. Reported by eliel on - #asterisk-dev. - - * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf - (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a - few potential deadlocks in cdr_sqlite3_custom. (also rename - sample config to .sample) Closes issue #11208, patch by Laureano. - -2007-11-09 16:00 +0000 [r89129] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c - (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c, - include/asterisk/hashtab.h (added), main/config.c: This is the - perhaps the biggest, boldest, most daring change I've ever - committed to trunk. Forgive me in advance any disruption this may - cause, and please, report any problems via the bugtracker. The - upside is that this can speed up large dialplans by 20 times (or - more). Context, extension, and priority matching are all fairly - constant-time searches. I introduce here my hashtables - (hashtabs), and a regression for them. I would have used the - ast_obj2 tables, but mine are resizeable, and don't need the - object destruction capability. The hashtab stuff is well tested - and stable. I introduce a data structure, a trie, for extension - pattern matching, in which knowledge of all patterns is - accumulated, and all matches can be found via a single traversal - of the tree. This is per-context. The trie is formed on the first - lookup attempt, and stored in the context for future lookups. - Destruction routines are in place for hashtabs and the pattern - match trie. You can see the contents of the pattern match trie by - using the 'dialplan show' cli command when 'core set debug' has - been done to put it in debug mode. The pattern tree traversal - only traverses those parts of the tree that are interesting. It - uses a scoreboard sort of approach to find the best match. The - speed of the traversal is more a function of the length of the - pattern than the number of patterns in the tree. The tree also - contains the CID matching patterns. See the source code comments - for details on how everything works. I believe the approach - general enough that any issues that might come up involving fine - points in the pattern matching algorithm, can be solved by just - tweaking things. We shall see. The current pattern matcher is - fairly involved, and replicating every nuance of it is difficult. - If you find and report problems, I will try to resolve than as - quickly as I can. The trie and hashtabs are added to the existing - context and exten structs, and none of the old machinery has been - removed for the sake of the multitude of functions that use them. - In the future, we can (maybe) weed out the linked lists and save - some space. - -2007-11-08 23:53 +0000 [r89124-89126] Jason Parker <jparker@digium.com> - - * /, main/say.c: Merged revisions 89125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600 - (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here.. - Issue 11203, fix described by vma. ........ - - * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be - used to check whether it is time to hangup a channel. Closes - issue #11202, patch by mnicholson - -2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: app_voicemail failed to build when - compiling with IMAP_STORAGE Now it does not. - - * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one - argument. Thanks to snuffy for pointing this out on IRC - -2007-11-08 21:27 +0000 [r89121] Joshua Colp <jcolp@digium.com> - - * funcs/func_env.c: Make func_env build again. - -2007-11-08 21:01 +0000 [r89120] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov - 2007) | 7 lines Rework of the commit I made yesterday to use the - already built-in ast_uri_decode function as opposed to my - home-rolled one. Also added comments. Thanks to oej for pointing - me in the right direction ........ - -2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_features.c: convert this code to a more efficient - idiom - -2007-11-08 18:49 +0000 [r89116-89117] Jason Parker <jparker@digium.com> - - * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch - by eliel - - * /, configs/cdr_adaptive_odbc.conf.sample, - configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600 - (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using - sample configuration files. Issue 11195, patch by eliel. ........ - -2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher <tlesher@digium.com> - - * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan - function and deprecate ReadFile. - - * channels/chan_features.c: Fix missed conversion to linkedlists - macro change - -2007-11-08 16:51 +0000 [r89112] Mark Michelson <mmichelson@digium.com> - - * /: Blocking changes from previous 1.4 commit - -2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo <rizzo@icir.org> - - * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored - anyways). - - * main/audiohook.c: use %d and cast to int instead of %zd for - size_t object, this helps portability. - - * channels/chan_unistim.c: initialize a variable to silence - compiler. The type of warnings emitted depends on the - optimization level, at the lower levels the compiler doesn't - always understand what the programmer has in mind. In this case I - could not understand it either. - -2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming <kpfleming@digium.com> - - * main/srv.c, /: Merged revisions 89105 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007) - | 2 lines fix a glaring bug in the new SRV record handling that - would cause incorrect weight sorting ........ - - * main/autoservice.c, main/frame.c, apps/app_meetme.c, - res/res_features.c, funcs/func_strings.c, main/devicestate.c, - res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, - codecs/codec_zap.c, res/res_jabber.c, main/indications.c, - main/astobj2.c, main/config.c, main/loader.c, main/cli.c, - main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c, - main/logger.c, main/app.c, main/image.c, res/res_speech.c, - main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c, - channels/chan_agent.c, utils/astman.c, apps/app_queue.c, - channels/iax2-parser.c, main/srv.c, - include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c, - main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c, - apps/app_voicemail.c: improve linked-list macros in two ways: - - the *_CURRENT macros no longer need the list head pointer - argument - add AST_LIST_MOVE_CURRENT to encapsulate the - remove/add operation when moving entries between lists - -2007-11-08 05:00 +0000 [r89104] Tilghman Lesher <tlesher@digium.com> - - * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) - | 2 lines Typo ........ - -2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 - lines Do not add a sip: to the beginning of the To URI unless - needed. (closes issue #10756) Reported by: goestelecom ........ - - * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 - lines Improve the devicestate logic for multiple devices. If any - are available then the extension is considered available. (closes - issue #10164) Reported by: nic_bellamy Patches: - sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) - ........ - - * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 - lines Add support for allowing one outgoing transaction. This - means if a response comes back out of order chan_sip will still - handle it. I dream of a chan_sip with real transaction support. - (closes issue #10946) Reported by: flefoll (closes issue #10915) - Reported by: ramonpeek (closes issue #9567) Reported by: - atca_pres ........ - - * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 - lines If callerid is configured in sip.conf use that for checking - the presence of an extension in the dialplan. (closes issue - #11185) Reported by: spditner ........ - -2007-11-07 23:47 +0000 [r89094] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) - | 7 lines The member refcount must be incremented, to avoid using - it after deallocation. A huge thanks go to lvl- for patiently - providing the necessary valgrind output that was necessary to - finding this problem of memory corruption. Reported by: lvl- - Patch by: tilghman Closes issue #11174 ........ - -2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: If imapfolder has been specified in - voicemail.conf, we should not connect to INBOX... ever. It may - not exist. (closes issue #11151, reported by selsky, patched by - me) - - * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov - 2007) | 6 lines This patch makes it possible for SIP phones to - dial extensions defined with '#' characters in extensions.conf - AND maintain their escaped characters when forming URI's (closes - issue #10681, reported by cahen, patched by me, code review by - file) ........ - -2007-11-07 22:09 +0000 [r89089] Steve Murphy <murf@digium.com> - - * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 - line In response to 10578, I just ran 1.4 thru valgrind; some of - the config leakage I've already fixed, but it doesn't hurt to - double check. I found and fixed leaks in res_jabber, cdr_tds, - pbx_ael. Nothing major, tho. ........ - -2007-11-07 17:45 +0000 [r89086] Joshua Colp <jcolp@digium.com> - - * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds - again. - -2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo <rizzo@icir.org> - - * Makefile: remove enter/exit comments when handling subdirectory. - If we really want them we can remove the --no-print-directory - - * main/loader.c: remove a debugging message which i forgot in. - - * Makefile: match changes in menuselect's Makefile - -2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher <tlesher@digium.com> - - * apps/app_playback.c: Suppress erroneous warnings on load. - Reported by: eliel Patch by: eliel Closes issue #11177 - - * /, configs/extensions.ael.sample: Merged revisions 89079 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) - | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch - by: eliel Closes issue #11178 ........ - - * channels/chan_zap.c, configs/zapata.conf.sample: Provide the - ability to directly manipulate the TON/NPI bits in the - dialstring. Reported by: thetatag Patch by: - thetatag/stevens/tilghman Closes issue #5331 - - * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy, - which provides FastAGI functionality for EAGI, while also - buffering the audio stream. Reported by: devil_slayer Patch by: - devil_slayer Closes issue #8921 - -2007-11-07 00:16 +0000 [r89076] Russell Bryant <russell@digium.com> - - * main/astmm.c: Fix another CLI command so it doesn't run the real - code when called for initialization. - -2007-11-07 00:04 +0000 [r89075] Mark Michelson <mmichelson@digium.com> - - * doc/tex/imapstorage.tex: Adding documentation regarding - imapfolder, imapgreetings, and greetingsfolder options in - voicemail.conf (closes issue #11133, reported by selsky, patched - by blitzrage) - -2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant <russell@digium.com> - - * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the - channel name as a prefix to the "agi debug" output. This makes - AGI debugging on busy systems much easier. (closes issue #10730) - Reported by: junky Patches: agi_debug_chan.diff uploaded by junky - (license 177) 20070923_10730.diff uploaded by mvanbaak (license - 7) - - * apps/app_meetme.c, CHANGES: Added the ability to do "meetme - concise" with the "meetme" CLI command. This extends the concise - capabilities of this CLI command to include listing all - conferences, instead of an addition to the other sub commands for - the "meetme" command. (closes issue #11078) Reported by: jthomas - Patches: meetme-concise.patch uploaded by jthomas (license 293) - -2007-11-06 23:08 +0000 [r89072] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Fix up some PBX logic that became broken. The code - would exit prematurely when it should have been collecting more - digits. (closes issue #11175) Reported by: pj - -2007-11-06 22:51 +0000 [r89071] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_jingle.c, channels/chan_phone.c, - codecs/codec_g722.c, main/frame.c, channels/chan_sip.c, - channels/chan_skinny.c, main/translate.c, channels/chan_h323.c, - main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h, - main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h: - Commit some cleanups to the format type code. - Remove the - AST_FORMAT_MAX_* types, as these are consuming 3 out of our - available 32 bits. - Add a native slin16 type, so that 16kHz - codecs can translate without losing resolution. (This doesn't - affect anything immediately, until another codec has wb support.) - -2007-11-06 22:36 +0000 [r89070] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the - queue strategy wrandom (closes issue #10942, reported and patched - by julianjm, documentation changes by me) - -2007-11-06 22:15 +0000 [r89069] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added - the S() and L() options to the MeetMe application. These are - pretty much identical to the S() and L() options to Dial(). They - let you set timeouts for the conference, as well as have warning - sounds played to let the caller know how much time is left, and - when it is running out. (closes issue #8030) Reported by: areski - Patches: meetme_timeout_timelimit_v2.patch uploaded by areski - (license 29) - -2007-11-06 22:05 +0000 [r89068] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Added CLI and manager commands for changing a - queue member's penalty (closes issue #9374, reported and - initially patched by wuwu, intermediate patch by eliel, and final - patch by me) - -2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add some more locking as well as API update - for libss7 for new transport types - -2007-11-06 21:08 +0000 [r89062] Steve Murphy <murf@digium.com> - - * /, main/config.c: Merged revisions 89036 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 - line closes issue #8786 - where the [catname](!) and - [catname](othercat1,othercat2,...) notation gets dropped across a - ConfigUpdate (or any other thing that would cause a config file - to be written). While I was at it, I also cleaned up some of the - destroy routines to free up comments, which was not being done. - Made sure the new struct I introduced is also cleaned up properly - at destruction time. My code handles multiple template - inclusions. Many thanks to ssokol for his patch, which, while not - literally used in the final merge, served as a foundation for the - fix. ........ - -2007-11-06 20:55 +0000 [r89057] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Remove native bridging check for DTMF based - transfers. Thanks to the last batch of RTP changes it is no - longer required for the media stream to go through Asterisk if - DTMF is going over signalling. It will simply reinvite back as - needed. (closes issue #11172) Reported by: ibc - -2007-11-06 20:32 +0000 [r89055] Mark Michelson <mmichelson@digium.com> - - * res/res_features.c: Instead of trying to callback a local channel - on a failed attended transfer, call the device that made the - transfer instead. This makes for much smoother calling back when - queues are involved. (closes issue #11155, reported by IPetrov) - Tremendous thanks to Russell for pulling me out of my block I was - having on this one - -2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 - Nov 2007) | 3 lines Fix init_classes() so that classes that - actually do have files loaded aren't treated as empty, and - immediately destroyed ... ........ - - * main/astmm.c: Fix the memory show allocations CLI command so that - it doesn't spew out all of the current memory allocations when - you start Asterisk, when the command's handler gets called for - initialization. - -2007-11-06 19:40 +0000 [r89051] Steve Murphy <murf@digium.com> - - * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in - OSX for a problem blitzrage found - -2007-11-06 19:23 +0000 [r89050] Olle Johansson <oej@edvina.net> - - * main/fskmodem.c: Formatting. Illegaly using some spare spaces - from Russell's space-bucket. - -2007-11-06 19:16 +0000 [r89049] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 - Nov 2007) | 2 lines We went to the trouble of creating a method - of tracking failed trylocks, then never turned it on (oops). - ........ - -2007-11-06 19:10 +0000 [r89048] Olle Johansson <oej@edvina.net> - - * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes - (preparing for SIP changes - adding TDD support to SIP) - -2007-11-06 19:10 +0000 [r89047] Jason Parker <jparker@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 - lines Correctly set the total number of channels from a zaptel - transcoder board. SPD-49, patch by Matthew Nicholson. ........ - -2007-11-06 19:04 +0000 [r89044] Mark Michelson <mmichelson@digium.com> - - * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c, - apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c, - apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c, - apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c, - apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c, - apps/app_softhangup.c, apps/app_waitforsilence.c, - channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c, - apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c, - apps/app_forkcdr.c, apps/app_waitforring.c, - apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c, - res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c, - apps/app_stack.c: "show application <foo>" changes for clarity. - (closes issue #11171, reported and patched by blitzrage) Many - thanks! - -2007-11-06 19:04 +0000 [r89043] Olle Johansson <oej@edvina.net> - - * /, main/tdd.c: Merged revisions 89042 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 - lines Bug fixes to tdd support in zaptel. ........ (Small changes - for trunk) - -2007-11-06 18:44 +0000 [r89041] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, include/asterisk/jabber.h, - channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle - to use TLS connections again. Closes issue #9972 - -2007-11-06 18:23 +0000 [r89038] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 - Nov 2007) | 11 lines If someone were to delete the files used by - an existing MOH class, and then issue a reload, further use of - that class could result in a crash due to dividing by zero. This - set of changes fixes up some places to prevent this from - happening. (closes issue #10948) Reported by: jcomellas Patches: - res_musiconhold_division_by_zero.patch uploaded by jcomellas - (license 282) Additional changes added by me. ........ - -2007-11-06 17:10 +0000 [r89034] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 - lines Make it so that if a peer is determined to be unreachable - using qualify their devicestate will report back unavailable. - (closes issue #11006) Reported by: pj ........ - -2007-11-06 17:05 +0000 [r89031] Luigi Rizzo <rizzo@icir.org> - - * main/loader.c: Fix embedding of modules on FreeBSD: the - constructor for the list of modules was run after the - constructors for the embedded modules (which appended entries to - the list). As a result, the list appeared empty when it was time - to use it. On linux the order of execution of constructor was - evidently different (it may depend on the ordering of modules in - the ELF file). This is only a workaround - there may be other - situations where the execution of constructors causes problems, - so if we manage to find a more general solution this workaround - can go away. - -2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged - revisions 88994 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 - lines Fix improbable but possible memory leaks in chan_zap. - (closes issue #11166) Reported by: eliel Patches: - chan_zap.c.patch uploaded by eliel (license 64) ........ - - * channels/chan_agent.c: Update chan_agent documentation. Change a - | to , as that is now the required way. (closes issue #11167) - Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel - (license 64) - -2007-11-06 15:01 +0000 [r88973] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_unistim.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Set up detection - of IP_PKTINFO in autoconf for chan_unistim - -2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant <russell@digium.com> - - * channels/chan_unistim.c: convert uses of LOG_DEBUG to use - ast_debug() - - * channels/chan_unistim.c, configs/unistim.conf.sample: Add - jitterbuffer support to chan_unistim. (closes issue #11168) - Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by - IgorG (license 20) - - * main/pbx.c, /, channels/busy.h, channels/ringtone.h, - include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | - 12 lines After seeing crashes related to channel variables, I - went looking around at the ways that channel variables are - handled. In general, they were not handled in a thread-safe way. - The channel _must_ be locked when reading or writing from/to the - channel variable list. What I have done to improve this situation - is to make pbx_builtin_setvar_helper() and friends lock the - channel when doing their thing. Asterisk API calls almost all - lock the channel for you as necessary, but this family of - functions did not. (closes issue #10923, reported by atis) - (closes issue #11159, reported by 850t) ........ - - * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 - Nov 2007) | 8 lines Remove some checks to see if locks are - initialized from the non-DEBUG_THREADS versions of the lock - routines. These are incorrect for a number of reasons: - It - breaks the build on mac. - If there is a problem with locks not - getting initialized, then the proper fix is to find that place - and fix the code so that it does get initialized. - If additional - debug code is needed to help find the problem areas, then this - type of things should _only_ be put in the DEBUG_THREADS - wrappers. ........ - -2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo <rizzo@icir.org> - - * channels/Makefile: explain that the host environment must be used - to build gentone; Remove unset variables, they would be - misleading. - - * Makefile: don't export variables that can be retrieved from - makeopts in child subdirs - -2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05 - Nov 2007) | 2 lines update comment to match the state of the code - ........ - -2007-11-05 23:31 +0000 [r88827] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 88826 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov - 2007) | 6 lines Reworked deadlock avoidance in __ast_read. - Restored audio to callback agents. (closes issue #11071, reported - by callguy, patched by me, tested by callguy and Ted Brown) - ........ - -2007-11-05 21:36 +0000 [r88770] Luigi Rizzo <rizzo@icir.org> - - * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level - Makefile. This too is used only in one place. - -2007-11-05 21:35 +0000 [r88769] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | - 8 lines When traversing the list of channel variables here in - transmit_invite(), the asterisk channel must be locked, as this - data may change at any time. (I have seen numerous reports of - crashes related to the handling of channel variables. There are a - couple of issues on the bug tracker related to it, but it has - also been noted on IRC and mailing lists. So, I am finding and - fixing some places where channel variables are handled - improperly.) ........ - -2007-11-05 21:27 +0000 [r88767] Luigi Rizzo <rizzo@icir.org> - - * Makefile, main/Makefile: Move the last instance of AST_LIBS to - the only place it is used, namely main/Makefile . I am unclear - where decisions on the build environment (CFLAGS, LDFLAGS, LIBS - and so on) should be made - right now they are split here and - there. As a first step in cleaning up this situation, i am trying - to at least collect all instances of each variable in one place. - -2007-11-05 21:23 +0000 [r88766] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | - 2 lines Fix up some indentation. ........ - -2007-11-05 20:50 +0000 [r88764] Luigi Rizzo <rizzo@icir.org> - - * Makefile.moddir_rules: comment out an unused variable. Remove it - in a few days if no problems arise. - -2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant <russell@digium.com> - - * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | - 7 lines Merge changes from - asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV - record support in Asterisk was broken. There was no guarantee on - what record Asterisk would choose to actually use. This set of - changes improves the situation by ensuring that Asterisk will - choose the highest priority record. ........ - - * main/channel.c, /: Merged revisions 88709 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | - 20 lines Merge the last bit of changes from - asterisk/team/russell/readq-1.4 The issue here is that the - channel frame readq handling got broken when the code was - converted to use the linked list macros. It caused corruption of - the list head and tail pointers. So, I fixed up the usage of the - linked list macros and in passing, simplified the code. I also - documented what the code is doing, as it was a bit difficult to - figure out at first. This bug showed itself with crashes showing - messed up head/tail pointers for the readq. However, there are a - couple of crashes that aren't quite as obvious, but I think may - be related. So, if your bug gets closed by this commit, but you - still have a problem, please reopen or create a new bug report. - (closes issue #10936) (closes issue #10595) (closes issue #10368) - (closes issue #11084) (closes issue #10040) (closes issue #10840) - ........ - -2007-11-05 19:22 +0000 [r88675] Luigi Rizzo <rizzo@icir.org> - - * Makefile: Cleanup the installation of samples, avoiding - repetitions. I am preserving the behaviour on *.adsi files, i.e. - overwrite anything there without making a backup. However I am - not sure that this is the intended behaviour. - -2007-11-05 18:52 +0000 [r88673] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 - lines If a SIP channel is put on hold multiple times do not keep - incrementing the onHold value. (closes issue #11085) Reported by: - francesco_r Tested by: blitzrage (closes issue #10474) Reported - by: acennami ........ - -2007-11-05 18:22 +0000 [r88653] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: Change wording to that suggested by MasterYoda - -2007-11-05 18:00 +0000 [r88652] Luigi Rizzo <rizzo@icir.org> - - * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned - output. - -2007-11-05 17:52 +0000 [r88651] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 88624 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | - 5 lines Fix up datastore handling in ast_do_masquerade(). The - code is intended to move any channel datastores from the old - channel to the new one. However, it did not use the linked list - macros properly to accomplish the task. The existing code would - only work if there was only a single datastore on the old - channel. ........ - -2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo <rizzo@icir.org> - - * Makefile: print messages when entering/leaving a directory so we - know where we are (sometimes it is obvious, sometimes it is not). - - * Makefile.moddir_rules: merge two rules with the same right hand; - document a bit what is done here. - -2007-11-05 17:21 +0000 [r88586] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 - (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config - structure on configuration failure. Issue 11163, patch by eliel. - ........ - -2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile.rules: use a variable name that actually indicates what - it is for - -2007-11-05 16:41 +0000 [r88553] Luigi Rizzo <rizzo@icir.org> - - * Makefile.rules: Put extra compiler flags into a variable so they - are not repeated too many times. On passing, add some comments - and fix indentation a bit. On passing, i suspect that the - following pattern is wrong %.eoo: %.o but in case it will be - fixed in a later commit. - -2007-11-05 16:30 +0000 [r88540] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007) - | 4 lines Don't check used pooled connections for connection - status, as it will cause issues for prepared queries. Reported - by: Nick Gorham (via -dev list) Patch by: tilghman ........ - -2007-11-05 15:15 +0000 [r88525] Luigi Rizzo <rizzo@icir.org> - - * main/db.c: remove a cygwin-specific function remap that does not - work. - -2007-11-05 13:11 +0000 [r88510] Joshua Colp <jcolp@digium.com> - - * channels/chan_unistim.c: Fix memory leaks and deadlocks in - chan_unistim. (closes issue #11158) Reported by: eliel Patches: - chan_unistim.c.patch uploaded by eliel (license 64) - -2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo <rizzo@icir.org> - - * /: block merging of not-applicable patch - - * main/channel.c, main/pbx.c, apps/app_meetme.c, - channels/chan_sip.c, res/res_features.c, main/utils.c, - channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify - the implementation and the API for stringfields; details and - examples are in include/asterisk/stringfields.h. Not applicable - to older branches except for 1.4 which will receive a fix for the - routines that free memory pools. - -2007-11-03 14:19 +0000 [r88437] Tilghman Lesher <tlesher@digium.com> - - * main/term.c: Revert commit #86119. Some users intentionally do - not want colorized terminals, so this was a misfeature. - -2007-11-03 04:55 +0000 [r88422] James Golovich <james@gnuinter.net> - - * main/db.c: Set CLI command to the correct name. Rev 85460 - introduced two 'database show' commands when this one should have - been 'database showkey' - -2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant <russell@digium.com> - - * channels/chan_unistim.c: fix some issues with crashing on unload, - when it didn't completely load cleanly - - * channels/chan_unistim.c: Convert the CLI commands to the new - format - - * pbx/pbx_lua.c: propagate the DECLINE return value back to the - loader - - * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not - present. - - * main/cli.c: Show the channel unique ID in the "show channel - concise" output (closes issue #11148, requested by falves11, - patched by me) - - * channels/chan_unistim.c (added), CREDITS, - configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt - (added): Merge the code from asterisk/team/group/chan_unistim: - This introduces a new channel driver, chan_unistim, that supports - the Unistim VoIP protocol for Nortel phones. The following models - have been confirmed to work: i2002, i2004 and i2050. (closes - issue #8864) Reported by: c_hans Patches: chan_unistim.patch - uploaded by c (license 304) ustm_no_conf.diff uploaded by junky - (license 177) Tested by: c_hans, dbowerman, math, junky, loloski - -2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 - lines Make subscribecontext behave as advertised. It will now - look for the presence of a hint in the given context (be it - subscribecontext or context). (closes issue #10702) Reported by: - slavon ........ - - * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 - lines If an INFO request within a dialog is received with a - content length of 0 simply send back a 200 OK. It is valid to do - this and the remote side is probably using it to make sure the - signalling is still alive. (closes issue #5747) Reported by: - chandi Patches: infofix-81430-1.patch uploaded by IgorG (license - 20) ........ - -2007-11-02 20:13 +0000 [r88327] Russell Bryant <russell@digium.com> - - * doc/tex/Makefile: Fix replacing the version number when it has a - '/' in it, like SVN-group-chan_unistim-r88326M-/trunk - -2007-11-02 17:34 +0000 [r88287] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90 - -2007-11-02 16:54 +0000 [r88284] Jason Parker <jparker@digium.com> - - * /, main/say.c: Merged revisions 88283 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500 - (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a - language to ast_fileexists, otherwise it may fail for anything - besides en Issue 11147, fix discovered by both citats and myself - (independently), with input from Corydon76 ........ - -2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: Add a few bytes on LUA - - * main/pbx.c, utils/build-extensions-conf.lua (added), - build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c - (added), configs/extensions.lua.sample (added), - include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of - doing extensions Reported by: mnicholson Patch by: mnicholson - Closes issue #11140 - - * main/config.c: Don't re-cache the filename, but check to see if - it already exists Reported by: jamesgolovich Patch by: - jamesgolovich Closes issue #11144 - - * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02 - Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy - Patch by: ys Closes issue #11143 ........ - - * main/pbx.c: 'h' extension doesn't execute past first priority - Reported by: dimas Patch by: dimas Closes bug #11146 - -2007-11-02 03:09 +0000 [r88197] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_odbc.c: Restore building under 64-bit platforms. - -2007-11-01 23:26 +0000 [r88184] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, configure, - include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in: - Remove traces of gnutls, since we no longer use/need it. - -2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Modify WaitExten to include an optional dialtone - Closes issue #10783 - - * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc - managed connections Closes issue #10614 - -2007-11-01 22:26 +0000 [r88166] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, - funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c, - apps/app_exec.c, apps/app_queue.c, apps/app_playback.c, - res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c, - apps/app_minivm.c, res/res_agi.c, main/logger.c, - pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c, - utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c, - apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c, - cdr/cdr_manager.c: This commits the performance mods that give - the priority processing engine in the pbx, a 25-30% speed boost. - The two updates used, are, first, to merge the - ast_exists_extension() and the ast_spawn_extension() where they - are called sequentially in a loop in the code, into a slightly - upgraded version of ast_spawn_extension(), with a few extra args; - and, second, I modified the substitute_variables_helper_full, so - it zeroes out the byte after the evaluated string instead of - demanding you pre-zero the buffer; I also went thru the code and - removed the code that zeroed this buffer before every call to the - substitute_variables_helper_full. The first fix provides about a - 9% speedup, and the second the rest. These figures come from the - 'PIPS' benchmark I describe in blogs, conf. reports, etc. - -2007-11-01 22:19 +0000 [r88164-88165] Jason Parker <jparker@digium.com> - - * /: Crap, accidentally copied the props. Thanks for pointing this - out mvanbaak. The odds are quite high that this will break - automerge on every team branch. - - * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber - to use openssl rather than gnutls. Closes issue #9972, patch by - phsultan. Copied from branch at - http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/ - -2007-11-01 17:25 +0000 [r88117] Tilghman Lesher <tlesher@digium.com> - - * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 - Nov 2007) | 2 lines Add some notes on using valgrind ........ - -2007-11-01 16:22 +0000 [r88079] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 - lines Make sure we set the poll fds to NULL after free()ing it. - Part of issue 11017, patch by tzafrir. ........ - -2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free() - to ast_free(). (No functional differences.) (closes issue #11138) - Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel - (license 64) chan_sip.c.patch uploaded by eliel (license 64) - - * utils/Makefile: Remove another copied source file on "make - clean". (closes issue #11137) Reported by: IgorG Patches: - addonclean-87971-1.patch uploaded by IgorG (license 20) - -2007-11-01 13:30 +0000 [r88027] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 - lines Fix up commit for my Zap channel with spies in Meetme fix. - (thanks Tony Mountifield!) ........ - -2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher <tlesher@digium.com> - - * main/utils.c: Conditionally free lock_info->thread_name to avoid - a useless warning Reported by: snuffy Patch by: snuffy Closes - issue #11125 - - * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to - pair calls of ast_malloc and ast_calloc Reported by: eliel Patch - by: eliel Closes issue #11135 - - * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed - by: tilghman Closes issue #11136 - -2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 - lines If a Zap channel contains a spy or a spy is added take it - out of the conference in kernel space and make it go through - Asterisk so the spy gets audio from both sides. (closes issue - #10060) Reported by: mparker ........ - - * main/pbx.c: Drop any more references to type in the Exception - dialplan function. (closes issue #11134) Reported by: blitzrage - Patches: exception_patch.txt uploaded by blitzrage (license 10) - -2007-10-31 21:23 +0000 [r87889-87909] Jason Parker <jparker@digium.com> - - * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500 - (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated - memory before returning. Issue 11131, patch by eliel. ........ - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 87906 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11130) (closes issue #11132) ........ r87906 | qwell | - 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try - to allocate memory that we're just going to re-allocate later - anyways. Issues 11130 and 11132, patch by eliel. ........ - - * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c, - formats/format_wav_gsm.c, res/res_musiconhold.c, - codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c, - formats/format_pcm.c, formats/format_h263.c, - formats/format_h264.c, formats/format_jpeg.c, - formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c, - codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, - formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c, - res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c, - formats/format_ogg_vorbis.c, res/res_ael_share.c, - formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c, - res/res_indications.c, codecs/codec_ilbc.c, - formats/format_g726.c, formats/format_g729.c: More changes to - change return values from load_module functions. (issue #11096) - Patches: codec_adpcm.c.patch uploaded by moy (license 222) - codec_alaw.c.patch uploaded by moy (license 222) - codec_a_mu.c.patch uploaded by moy (license 222) - codec_g722.c.patch uploaded by moy (license 222) - codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch - uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy - (license 222) codec_lpc10.c.patch uploaded by moy (license 222) - codec_speex.c.patch uploaded by moy (license 222) - codec_ulaw.c.patch uploaded by moy (license 222) - codec_zap.c.patch uploaded by moy (license 222) - format_g723.c.patch uploaded by moy (license 222) - format_g726.c.patch uploaded by moy (license 222) - format_g729.c.patch uploaded by moy (license 222) - format_gsm.c.patch uploaded by moy (license 222) - format_h263.c.patch uploaded by moy (license 222) - format_h264.c.patch uploaded by moy (license 222) - format_ilbc.c.patch uploaded by moy (license 222) - format_jpeg.c.patch uploaded by moy (license 222) - format_ogg_vorbis.c.patch uploaded by moy (license 222) - format_pcm.c.patch uploaded by moy (license 222) - format_sln.c.patch uploaded by moy (license 222) - format_vox.c.patch uploaded by moy (license 222) - format_wav.c.patch uploaded by moy (license 222) - format_wav_gsm.c.patch uploaded by moy (license 222) - res_adsi.c.patch uploaded by eliel (license 64) - res_ael_share.c.patch uploaded by eliel (license 64) - res_clioriginate.c.patch uploaded by eliel (license 64) - res_convert.c.patch uploaded by eliel (license 64) - res_indications.c.patch uploaded by eliel (license 64) - res_musiconhold.c.patch uploaded by eliel (license 64) - res_smdi.c.patch uploaded by eliel (license 64) - res_speech.c.patch uploaded by eliel (license 64) - -2007-10-31 18:53 +0000 [r87888] Steve Murphy <murf@digium.com> - - * /: Merged revisions 87849 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1 - line closes issue #11108 -- where the 'dialplan save' cli command - saves a file where the semicolon is not escaped. Fixed this; User - also wanted comments to be preserved across dialplan save, but - this is impossible at this point in time, because comments are - not stored in the dialplan. They are 'compiled' out of - extensions.conf. The only way to preserve those comments is to - use the config file reader/writer that the GUI uses to allow - online user edits. extensions.conf is first and foremost, a - config file, and is read in by the normal config-file reading - routines. Then, it is processed into a dialplan (context/exten - structs). (in the case of trunk, tho, no mods needed to be made - -- works OK there -- just make sure you use ',' to sep app args!) - ........ - -2007-10-31 18:09 +0000 [r87854] Tilghman Lesher <tlesher@digium.com> - - * Makefile, /: Merged revisions 87852 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007) - | 2 lines Create samples for ALL of the available options in - asterisk.conf ........ - -2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: Add volume adjustment in. - - * apps/app_mixmonitor.c: Restore operation of the option that only - writes when the channel is bridged. - - * apps/app_chanspy.c: Add volume adjustment to spy audiohook in - app_chanspy. - -2007-10-31 16:13 +0000 [r87817] Tilghman Lesher <tlesher@digium.com> - - * CREDITS: Formatting cleanups, remove obsolete contributions - (modules no longer in Asterisk), and obfuscate email addresses - enough to stop most spam harvesters. - -2007-10-31 16:07 +0000 [r87815] Joshua Colp <jcolp@digium.com> - - * include/asterisk/channel.h: Remove old whisper remnants from - channel.h - -2007-10-31 15:46 +0000 [r87811] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Optimize pbx_substitute_variables - -2007-10-31 04:20 +0000 [r87776] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 - line Included some verbage in the check_includes func, to inform - the user that included contexts that have no match in the AEL, - might be OK, as AEL cannot check in the extensions.conf or the - in-memory contexts, as they may not be there at the time of the - check. ........ - -2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30 - Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD - Reported by: ys Fixed by: ys Closes issue #11116 ........ - - * apps/app_exec.c: If no '?' is found in the arguments, don't - attempt to continue. Reported by: blitzrage Fixed by: tilghman - Closes issue #11111 - -2007-10-30 21:22 +0000 [r87687] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | - 11 lines Merge the changes from team/russell/iax2_poke_fix and - iax2-poke-fix-trunk There was a race condition related to the - handling of POKEing peers. Essentially, a reference to a peer is - held by the scheduler when there are pending callbacks, but the - reference count didn't reflect it. So, it was possible for a peer - to hit a reference count of zero and have its destructor begin to - be called at the same time that the scheduler thread ran a POKE - related callback. If that happened, a crash would likely occur. - (closes issue #11082, closes issue #11094) ........ - -2007-10-30 20:30 +0000 [r87626-87651] Jason Parker <jparker@digium.com> - - * /, channels/Makefile: Merged revisions 87650 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 - line Only try to clean out h323/ if the h323/Makefile exists. - ........ - - * main/pbx.c: Update documentation to give an example of how to use - the return status of RaiseException Closes issue #11117, patch by - blitzrage (yay blitzrage) - -2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c: The priority gets incremented after raising an - exception, so the priority should be set to 0 - - * main/pbx.c: Jumped the gun a bit in the RaiseException app. It - would always return -1 since it checked for the existence of - something that will never exist. - -2007-10-30 16:15 +0000 [r87572] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 87571 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4 - lines Add two more checks before printing out a warning message - about bridging. If either channel has hungup of course the bridge - will have failed. (closes issue #10009) Reported by: dimas - ........ - -2007-10-30 15:47 +0000 [r87568] Jason Parker <jparker@digium.com> - - * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500 - (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris. - Issue 11113, patch by snuffy. ........ - -2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming <kpfleming@digium.com> - - * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to - fix #10995, I found all kinds of cruft in this Makefile. It - should all be gone now, and as a side effect hashtest2 now builds - with --enable-dev-mode enabled without a host of errors - - * agi/Makefile, utils/Makefile, codecs/g722/Makefile, - main/editline/Makefile.in, Makefile.moddir_rules, - codecs/ilbc/Makefile, codecs/lpc10/Makefile, - main/db1-ast/Makefile: clean up assembler and preprocessor files - if they are here too - - * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs, - codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main, - codecs/gsm, main/minimime, pbx, res, channels: ignore - preprocessor and assembler files if they are present - - * Makefile, /: Merged revisions 87460 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007) - | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is - already there (used when debugging preprocessor issues) because - the compiler will whine about each compile command ........ - -2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: If a caller is listen-only, then don't bother - with doing talker detection. (closes issue #10911, reported by - junky, patched by me) - - * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) | - 5 lines Add some more details to the output of "core show locks". - When a thread is waiting for a lock, this will now show the - details about who currently has it locked. (inspired by issue - #11100) ........ - -2007-10-29 20:13 +0000 [r87395] Mark Michelson <mmichelson@digium.com> - - * UPGRADE.txt, apps/app_queue.c: Adding the more flexible - QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function. - A deprecation notice will be issued the first time - QUEUE_MEMBER_COUNT is used. - -2007-10-29 20:02 +0000 [r87394] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Drop the RTCP Read too short message to debug. There - are some phones out there that send a sort of keep alive packet - in the RTCP that trigger this every 5 seconds. - -2007-10-29 19:56 +0000 [r87393] Jason Parker <jparker@digium.com> - - * apps/app_record.c: Make sure we set flags to a 0 value before - trying to use it. Pointed out by seanbright while I was debugging - issue 11109. - -2007-10-29 19:47 +0000 [r87392] Russell Bryant <russell@digium.com> - - * /, main/astmm.c: Merged revisions 87373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) | - 5 lines Remove a lock that doesn't make any sense. The regions - lock needs to be held when traversing the list of allocated - chunks so that they can be printed out to the CLI. (Thanks to - eliel on #asterisk-dev for pointing this out!) ........ - -2007-10-29 17:22 +0000 [r87343] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 - lines Fix issue where if both sides of the dialog cancelled the - dialog at the same time chan_sip could kepe retransmitting a - response for no reason. (closes issue #9566) Reported by: - atca_pres Patches: bug9566.patch uploaded by oej ........ - -2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes - issue #11105) Reported by: eliel Patches: app_voicemail.c.patch - uploaded by eliel (license 64) - - * channels/chan_misdn.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Add autoconf - checks for extra suppserv definitions that are not present in - releases yet. chan_misdn should now build against the latest - release. (closes issue #11103) Reported by: IgorG - - * /, main/utils.c: Merged revisions 87294 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6 - lines Fix issue with ast_unescape_semicolon going into an endless - loop. (closes issue #10550) Reported by: ramonpeek Patches: - unescape-85177-1.patch uploaded by IgorG (license 20) ........ - -2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_dialgroup.c (added): Add a simple dialgroup function. - By taking one of the simpler uses of Queue away from Queue, we - simplify the lives of people who do not need all the bells and - whistles. Also, this is part of the functions that people need to - reimplement Queue in the dialplan, as a set of logic, rather than - as a single app with hundreds of options. - - * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c, - funcs/func_realtime.c: Merged revisions 87262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007) - | 7 lines Add autoservice to several more functions which might - delay in their responses. Also, make sure that func_odbc - functions have a channel on which to set variables. Reported by - russell Fixed by tilghman Closes issue #11099 ........ - -2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the configure script for the last - libss7 API change - - * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists - before attempting to start or stop channel autoservice in - func_lock and func_shell. - -2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add Circuit Group Queury message code - - * channels/chan_zap.c: Make sure we turn on the DSP when we answer - the call - -2007-10-26 22:21 +0000 [r87217] Mark Michelson <mmichelson@digium.com> - - * CHANGES: Forgot to update CHANGES when I committed the linear - queue strategy. Thank you Russell, for pointing this out! - -2007-10-26 21:37 +0000 [r87202] Jason Parker <jparker@digium.com> - - * channels/chan_local.c, channels/chan_zap.c, - channels/chan_agent.c, channels/chan_features.c, - res/res_crypto.c, res/res_realtime.c, res/res_monitor.c: - Correctly use defined return values in (some) load_module - functions. (issue #11096) Patches: chan_agent.c.patch uploaded by - eliel (license 64) chan_local.c.patch uploaded by eliel (license - 64) chan_features.c.patch uploaded by eliel (license 64) - chan_zap.c.patch uploaded by eliel (license 64) - res_monitor.c.patch uploaded by eliel (license 64) - res_realtime.c.patch uploaded by eliel (license 64) - res_crypto.c.patch uploaded by eliel (license 64) - -2007-10-26 17:39 +0000 [r87187] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c, - res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c, - res/ael/ael.tab.h, utils/ael_main.c, - pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex, - utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged - revisions 87168 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 - line closes issue #11086 where a user complains that references - to following contexts report a problem; The problem was REALLy - that he was referring to empty contexts, which were being - ignored. Reporter stated that empty contexts should be OK. I - checked it out against extensions.conf, and sure enough, empty - contexts ARE ok. So, I removed the restriction from AEL. This, - though, highlighted a problem with multiple contexts of the same - name. This should be OK, also. So, I added the extend keyword to - AEL, and it can preceed the 'context' keyword (mixed with - 'abstract', if nec.). This will turn off the warnings in AEL if - the same context name is used 2 or more times. Also, I now call - ast_context_find_or_create for contexts now, instead of just - ast_context_create; I did this because pbx_config does this. The - 'extend' keyword thus becomes a statement of intent. AEL can now - duplicate the behavior of pbx_config, ........ - -2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: Added queue - strategy "linear". This strategy is useful for those who always - wish for their phones to be rung in a specific order. (closes - issue #7279, reported and initially patched by diLLec, patch - reworked by me) - - * configs/queues.conf.sample: Remove information about the - roundrobin strategy from trunk's queues.conf.sample since it no - longer exists - -2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007) - | 7 lines The addition of autoservice to func_curl additionally - made func_curl dependent on the existence of a channel, with no - real reason. This should make func_curl once again work without a - channel. Reported by jmls. Fixed by tilghman. Closes issue #11090 - ........ - - * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c, - main/app.c: Use the same delimited character as the FILTER - function in FIELDQTY and CUT. - -2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, /, include/asterisk/linkedlists.h: Merged - revisions 87069 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) - | 2 lines appending one list to another should leave the first - list empty, and not require the user to do that ........ - -2007-10-25 18:59 +0000 [r87040] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Add support for a muted user to request to - talk. The '2' option in the user menu will adjust this status if - a user is muted. The talk request status will be reflected in the - CLI commands as well as the manager interface. (closes issue - #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded - by imesper (license 275) - -2007-10-25 16:21 +0000 [r87024] Steve Murphy <murf@digium.com> - - * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c: - closes issue #11045 - each file needs to define - ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the - compiler flags; the problem was that the fixes were getting made - to the generated .c file, and erased the next time someone - regenerated that file from the corresponding .y or .flex file. - Moral of story: keep your eyes open and make mods to the .y (or - flex input file) and re-run bison (or flex) as the Makefile - directs for that file, and then check in both. Also, - res_config_sqlite was kinda missed, and has the same issue. - -2007-10-24 21:26 +0000 [r86985] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: Adding the general - option "shared_lastcall" to queues so that a member's wrapuptime - may be used across multiple queues. (closes issue #9777, reported - and patched by eliel) - -2007-10-24 20:59 +0000 [r86983] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500 - (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname - configuration option. Simplify code slightly in the process. - Issue 11079, reported by ddv2005 ........ - -2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2, - pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c, - pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue - #11005, where #include uses the current dir instead of the config - dir (/etc/asterisk) for relative path includes for AEL - - * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1 - line closes issue #11037 -- unable to specify app:spec in hint - arguments ........ - - * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1 - line closes issue #11052 -- where nothing after the ? will allow - un-initialized variable values to corrupt and crash asterisk on - 64-bit platforms ........ - - * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1 - line This should get rid of a really, really irritating warning - generated by some 64-bit platforms from libc, where free(0) is - frowned upon ........ - - * /, main/Makefile: Merged revisions 86881 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 - line this update to Makefile corrects how ast_expr2f.c should be - generated ........ - -2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant <russell@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22 - Oct 2007) | 9 lines If lock tracking is not enabled, then we can - not attempt to log any mutex failures. If so, we could end up in - infinite recursion. The only lock that is affected by this is a - mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue - #11044) Reported by: ys Patches: lock.h.diff uploaded by ys - (license 281) ........ - - * apps/app_playback.c: Convert some spaces to tabs and make it so - the CLI command is only registered once instead of 3 times. - (closes issue #11053) Reported by: seanbright Patches: - app_playback.patch uploaded by seanbright (license 71) - -2007-10-22 20:05 +0000 [r86820] Jason Parker <jparker@digium.com> - - * main/udptl.c, channels/chan_local.c, main/frame.c, - res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, - main/astobj2.c, main/config.c, main/cli.c, - channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, - channels/chan_alsa.c, main/db.c, main/pbx.c, - channels/chan_agent.c, channels/iax2-provision.c, - apps/app_playback.c, channels/chan_misdn.c, - channels/chan_features.c, res/res_indications.c, - pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, - res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, - main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, - res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, - main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, - res/res_agi.c, apps/app_minivm.c, main/logger.c, - res/res_realtime.c, main/image.c, apps/app_rpt.c, - channels/chan_mgcp.c, res/res_clioriginate.c, - res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, - channels/chan_sip.c, res/res_limit.c, main/translate.c, - res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, - apps/app_queue.c, channels/chan_oss.c, main/rtp.c, - channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, - channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, - funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to - AST_CLI_DEFINE, since the former didn't make much sense - -2007-10-22 17:40 +0000 [r86790] Tilghman Lesher <tlesher@digium.com> - - * /, main/astmm.c: Merged revisions 86787 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007) - | 2 lines Minor FreeBSD build fix ........ - -2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 - lines After reading online I have confirmed that Record-Route - headers should be copied to 1xx responses as well. (closes issue - #10113) Reported by: makoto ........ - - * /, apps/app_controlplayback.c: Merged revisions 86754 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4 - lines Make sure res is a positive value before performing the - check to determine whether the user stopped it or not. (closes - issue #11023) Reported by: cfc ........ - -2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 86750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | - 8 lines Don't leak a frame in the case that an END frame is - received and the time since the BEGIN is less than that of the - defined minimum DTMF duration. (closes issue #11051) Reported by: - casper Patches: channel.c.86664.diff uploaded by casper (license - 55) ........ - - * channels/chan_zap.c: There is a really fun game that you can play - before committing code, and it's called "make". :) - - * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22 - Oct 2007) | 4 lines Update the static mutex initializer to - include the initialization of the internal mutex used to protect - the lock debugging data. (closes issue #11044, patch suggested by - Ivan) ........ - -2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval - defaulting to something other than 'never' doesn't seem to - accomplish any good and causes problems for plenty of people... - -2007-10-22 14:58 +0000 [r86696] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct - 2007) | 5 lines Account for the fact that sometimes headers may - be terminated with \r\n instead of just \n (closes issue #11043, - reported by yehavi) ........ - -2007-10-22 14:56 +0000 [r86695] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c: merging patches that don't compile is bad... - mmkay? - -2007-10-22 14:28 +0000 [r86631-86664] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 86663 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6 - lines Move log message to before the frame it references is - freed. (closes issue #11050) Reported by: slavon Patches: - channel.c.86662.diff uploaded by casper (license 55) ........ - - * /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6 - lines Fix tab completion for dundi show peer. (closes issue - #11041) Reported by: jsmith Patches: - asterisk-dundicomplete.diff.txt uploaded by jamesgolovich - (license 176) ........ - - * /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct - 2007) | 6 lines Fixes for building under OpenSolaris. (closes - issue #11047) Reported by: snuffy Patches: 11047-fixes.diff - uploaded by snuffy (license 35) ........ - -2007-10-22 10:18 +0000 [r86616-86617] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 86598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) | - 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if - the dialplan does not match after an overlap call. Also added - out_cause=1 ........ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - started to add some basic support for supplementary services like - CallForwarding and so forth - -2007-10-21 22:52 +0000 [r86585] Russell Bryant <russell@digium.com> - - * /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged - revisions 85532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) | - 8 lines Properly handle the case where read() may return the text - for more than one CLI command at once for a remote console. - (closes issue #10888) Reported by: jamesgolovich Patches: - asterisk-climultiple.diff.txt uploaded by jamesgolovich (license - 176) ........ - -2007-10-20 19:56 +0000 [r86572] Matthew Fredrickson <creslin@digium.com> - - * configs/zapata.conf.sample: Improved comments and organization - for zapata.conf (#10904) - -2007-10-19 18:46 +0000 [r86549] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add better support for blocking and - unblocking of CICs (#10965) - -2007-10-19 18:29 +0000 [r86534-86536] Jason Parker <jparker@digium.com> - - * main/udptl.c, channels/chan_local.c, main/frame.c, - res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, - main/astobj2.c, main/config.c, main/cli.c, - channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, - channels/chan_alsa.c, main/db.c, main/pbx.c, - channels/chan_agent.c, channels/iax2-provision.c, - apps/app_playback.c, channels/chan_misdn.c, - channels/chan_features.c, res/res_indications.c, - pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, - res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, - main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, - res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, - main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, - res/res_agi.c, apps/app_minivm.c, main/logger.c, - res/res_realtime.c, main/image.c, apps/app_rpt.c, - channels/chan_mgcp.c, res/res_clioriginate.c, - res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, - channels/chan_sip.c, res/res_limit.c, main/translate.c, - res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, - apps/app_queue.c, channels/chan_oss.c, main/rtp.c, - channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, - channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, - funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue - #11039, as suggested by seanbright. - - * channels/chan_usbradio.c, res/res_config_pgsql.c, - channels/chan_misdn.c, channels/chan_h323.c, - res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c, - res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes - to NEW_CLI. Also fixes a few cli messages and some minor - formatting. (closes issue #11001) Reported by: seanbright - Patches: newcli.1.patch uploaded by seanbright (license 71) - newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch - uploaded by seanbright (license 71) newcli.5.patch uploaded by - seanbright (license 71) newcli.6.patch uploaded by seanbright - (license 71) newcli.7.patch uploaded by seanbright (license 71) - -2007-10-19 16:40 +0000 [r86470-86503] Joshua Colp <jcolp@digium.com> - - * /, main/app.c: Merged revisions 86502 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4 - lines When returning a DTMF digit from ast_control_streamfile - cast it as a char so that 0 does not overlap with the success - return code. (closes issue #11023) Reported by: cfc ........ - - * /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 - lines Fix two issues with domains and transfers. If a port was - given in the hostname it was treated as part of the hostname. If - domains were configured but external domains were not enabled all - transfers would be considered remote. (closes issue #11027) - Reported by: ramonpeek Patches: 11027-1.diff uploaded by - ramonpeek (license 266) ........ - - * /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 - lines Set port number in received as information for - registrations as well. (closes issue #11028) Reported by: brad-x - ........ - -2007-10-19 01:56 +0000 [r86439] TransNexus OSP Development <support@transnexus.com> - - * apps/app_osplookup.c: Fixed a buffer size issue. - -2007-10-18 22:03 +0000 [r86407-86408] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 86405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500 - (Thu, 18 Oct 2007) | 4 lines Add documentation for options in - asterisk.conf Issue 11029, patch by eserra ........ - -2007-10-18 18:40 +0000 [r86350] Mark Michelson <mmichelson@digium.com> - - * channels/chan_zap.c: Fixing a segfault from tab-completing a "zap - restart" CLI command. (patch made by seanbright, pointed out in - #asterisk-dev on IRC) - -2007-10-18 18:06 +0000 [r86331] Russell Bryant <russell@digium.com> - - * main/channel.c, /, include/asterisk/channel.h: Merged revisions - 86330 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | - 10 lines The channel needs to stay locked while running timer - callbacks, as they access and modify channel data that may change - elsewhere. I went through every timer callback in the source tree - to make sure that none of them did any additional locking that - could introduce deadlocks, and all is well. (closes issue #10765) - Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff - uploaded by Ivan (license 229) ........ - -2007-10-18 17:40 +0000 [r86298-86329] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 86328 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct - 2007) | 5 lines If a non-existent file is specified to be played - either as a periodic announcement or as a hold/position - announcement, the caller would be kicked out of the queue. No - longer does this happen. ........ - - * apps/app_queue.c: Changed some spaces to tabs - -2007-10-18 15:57 +0000 [r86297] Russell Bryant <russell@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) | - 3 lines Execute the RELEASE operation on transcoder channels in - the destroy callback. (patch from jsloan) ........ - -2007-10-18 07:23 +0000 [r86277-86278] Tilghman Lesher <tlesher@digium.com> - - * main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue - #10784 - - * res/res_musiconhold.c: On reload, re-read the files in the - specified moh directory (closes issue #10536) - -2007-10-18 04:41 +0000 [r86238] Russell Bryant <russell@digium.com> - - * /, main/utils.c: Merged revisions 86237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) | - 9 lines Revert a change that I made for issue #10979 which, as - has been pointed out to me in issue #11018, doesn't really make - sense. There is no reason to have the base64 decode function - force a '\0' terminated buffer, when the result is almost always - binary, anyway. In fact, this caused some breakage, as some code - in res_crypto passed in a buffer exactly the right size to get - its binary result, which got stomped on by this patch. (closes - issue #11018, reported by dimas) ........ - -2007-10-17 21:41 +0000 [r86208] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 86202 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct - 2007) | 6 lines Changing the strategy field of the call_queue - struct to be signed instead of unsigned, since the code attempts - to set the strategy to -1 if you specify a bogus strategy. While - this isn't a huge issue in 1.4, it could be a problem for someone - who, say, tries to use the roundrobin strategy in trunk (despite - all the deprecation warnings in 1.4). ........ - -2007-10-17 21:16 +0000 [r86195-86197] Tilghman Lesher <tlesher@digium.com> - - * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify - some preprocessor logic by using #elif - - * CHANGES, configs/meetme.conf.sample: Document the changes made - earlier today to meetme - -2007-10-17 20:06 +0000 [r86180-86182] Steve Murphy <murf@digium.com> - - * utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and - then, I noticed the clicompat stuff. - - * utils/check_expr.c: more stub routines to allow linkage in - stand-alone environment, with thread debugs turned on - - * utils/hashtest2.c: more stub routines to allow linkage in - stand-alone environment, with thread debugs turned on - -2007-10-17 18:01 +0000 [r86150] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | - 8 lines If Asterisk is in the middle of shutting down, respond to - OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: - eserra Patches: sip-options-503.patch uploaded by eserra (license - 45) ........ - -2007-10-17 17:06 +0000 [r86119] Tilghman Lesher <tlesher@digium.com> - - * main/term.c: Support color on certain platforms, even when - started at boot (before TERM is set) Closes issue #9048 - -2007-10-17 17:00 +0000 [r86118] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4 - lines Whoops, forgot to remove the original sip_scheddestroy. - (closes issue #11010) Reported by: vadim ........ - -2007-10-17 16:09 +0000 [r86104] Jason Parker <jparker@digium.com> - - * channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow - chan_usbradio to compile again. Closes issue #11014, patch by - seanbright. - -2007-10-17 15:39 +0000 [r86079] Tilghman Lesher <tlesher@digium.com> - - * /, main/asterisk.c: Merged revisions 86066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007) - | 3 lines When runuser/rungroup is specified, a remote console - could only be attained by root (Closes issue #9999) ........ - -2007-10-17 15:30 +0000 [r86067] Joshua Colp <jcolp@digium.com> - - * channels/chan_usbradio.c: Change dependency for chan_usbradio to - asound. Let's keep everything uniform. (closes issue #11013) - Reported by: seanbright - -2007-10-17 15:13 +0000 [r86065] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: Enhancements to realtime (closes issue #9609) - -2007-10-17 15:09 +0000 [r86064] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4 - lines Don't schedule dialog destruction if a MESSAGE is received - using an existing dialog. (closes issue #11010) Reported by: - vadim ........ - -2007-10-16 23:36 +0000 [r86029-86033] Mark Michelson <mmichelson@digium.com> - - * /, configs/queues.conf.sample: Merged revisions 86032 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct - 2007) | 3 lines Since monitor-join is deprecated now, remove the - example from the sample queues.conf file ........ - - * apps/app_queue.c: Removed the monitor-join option. If one wishes - to mix audio, they should instead use monitor-type=mixmonitor. - (related to issue #10885) - -2007-10-16 22:36 +0000 [r85995-85998] Russell Bryant <russell@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16 - Oct 2007) | 1 line really picky formatting tweak ... ........ - - * /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 - Oct 2007) | 16 lines Some locking errors exposed the fact that - the lock debugging code itself was not thread safe. How ironic! - Anyway, these changes ensure that the code that is accessing the - lock debugging data is thread-safe. Many thanks to Ivan for - finding and fixing the core issue here, and also thanks to those - that tested the patch and provided test results. (closes issue - #10571) (closes issue #10886) (closes issue #10875) (might close - some others, as well ...) Patches: (from issue #10571) - ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license - 229) - a few small changes by me ........ - -2007-10-16 21:51 +0000 [r85959-85992] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing the build. - - * apps/app_read.c: Fixing app_read so that if a timeout of less - than 1 ms is specified, assume that 1 ms is desired. (closes - issue #11000, reported and patched by michael-fig, with a warning - line added by me) - - * /, apps/app_queue.c: Merged revisions 85958 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct - 2007) | 5 lines Trying to remove a non-dynamic queue member via - dynamic means can lead to some interesting (read nasty) - situations. This patch clears up the issue by making only dynamic - queue members removable via dynamic methods. ........ - -2007-10-16 20:55 +0000 [r85957] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Don't hangup the call for SS7 if we get an - alarm - -2007-10-16 20:32 +0000 [r85944] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't - improperly memset() over an ast_str. This was leftover from - before it got changed to use ast_str. (closes issue #11003, - reported by pj) (closes issue #10770, reported by yehavi) - (patched by me) - -2007-10-16 19:47 +0000 [r85943] Tilghman Lesher <tlesher@digium.com> - - * /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16 - Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z - gnu extension to strftime) Reported and fixed by jcmoore Closes - issue #11002 ........ - -2007-10-16 19:12 +0000 [r85897] Russell Bryant <russell@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) | - 2 lines Remove a pointless lock. ........ - -2007-10-16 16:40 +0000 [r85853-85883] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix IMAP compilation error. (closes issue - #10986, reported and patched by snuffy) - - * /: Blocking changes from previous commit - -2007-10-16 15:15 +0000 [r85819-85851] Joshua Colp <jcolp@digium.com> - - * /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4 - lines Check to make sure a value has been given to the VMCOUNT - dialplan function. (closes issue #10996) Reported by: marsosa - ........ - - * main/threadstorage.c: Permit building under DEBUG_THREADLOCALS. - Thanks snuff. - - * /, main/threadstorage.c: Merged revisions 85818 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6 - lines Fix memory allocation issue in threadstorage. (closes issue - #10995) Reported by: snuffy Patches: new-patch.diff uploaded by - snuffy (license 35) ........ - -2007-10-16 10:38 +0000 [r85777-85787] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help - output - - * channels/chan_jingle.c: Added two CLI functions, taken from - chan_gtalk : - jingle reload ; - jingle show channels. - - * channels/chan_jingle.c: Make an audio path under the following - call configuration : SIP Phone 1 --- [chan_sip]Asterisk - 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP - Phone 2 Modifications : - set bridge type to partial ; - process - media candidates from the remote peer properly. Now we have - Jingle audio, at least between two Asterisk Jingle clients. - -2007-10-15 23:20 +0000 [r85764] Jason Parker <jparker@digium.com> - - * configs/dundi.conf.sample, channels/chan_sip.c, - channels/chan_h323.c, main/acl.c, UPGRADE.txt, - channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c, - channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new - tos config format. Remove old unused defines for old style. - Closes issue 10860, patch by IgorG. - -2007-10-15 21:11 +0000 [r85718-85721] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 85720 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) | - 3 lines Ensure that no pending state changes are leaked when the - device state change thread gets stopped on module unload. - ........ - - * /, main/say.c: Merged revisions 85686 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) | - 7 lines Add a small fix for the tw version of saying dates. - (closes issue #7827) Reported by: sharkey Patches: say.nits.patch - uploaded by sharkey (license 172) ........ - -2007-10-15 20:16 +0000 [r85685] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 85684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500 - (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config' - target. Do not try to run chkconfig or similar if using DESTDIR. - Issue 10938, patch by cabal95. ........ - -2007-10-15 20:09 +0000 [r85648-85683] Russell Bryant <russell@digium.com> - - * doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the - channel var docs - - * res/res_features.c, CHANGES: Added support for reading the - TOUCH_MONITOR_PREFIX channel variable. It allows you to configure - a prefix for auto-monitor recordings. (closes issue #6353) - Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded - by ivanfm (original patch) - updated patch: - 6353-touch_monitor_prefix.diff uploaded by qwell (license 4) - - * /, main/utils.c: Merged revisions 85649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) | - 2 lines Be pedantic about handling memory allocation failure. - ........ - - * /, main/utils.c: Merged revisions 85647 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) | - 5 lines The loop in the handler for the "core show locks" could - potentially block for some amount of time. Be a little bit more - careful and prepare all of the output in an intermediary buffer - while holding a global resource. Then, after releasing it, send - the output to ast_cli(). ........ - -2007-10-15 17:51 +0000 [r85633] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_strings.c: Document my changes from Friday - -2007-10-15 16:59 +0000 [r85605] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | - 6 lines Make the default for the srvlookup option to be yes. It - doesn't really make sense for it to default to off. The default - configuration file has it on, and proper RFC behavior, as - indicated by a comment in the code, is for it to be on. So, let's - have it on by default to make lives easier. (closes issue #10954, - suggested by jtodd) ........ - -2007-10-15 16:41 +0000 [r85578] Joshua Colp <jcolp@digium.com> - - * /, configs/features.conf.sample: Merged revisions 85571 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4 - lines Document that DTMF based features only work when two - channels are bridged together. (closes issue #10773) Reported by: - pbayley ........ - -2007-10-15 16:36 +0000 [r85562] Russell Bryant <russell@digium.com> - - * /, include/asterisk/strings.h: Merged revisions 85561 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) | - 4 lines Make a few changes so that characters in the upper half - of the ISO-8859-1 character set don't get stripped when reading - configuration. (closes issue #10982, dandre) ........ - -2007-10-15 16:23 +0000 [r85560] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 85559 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 - lines Bring both DTMF begin and end frames up through to the core - for DTMF feature handling. (closes issue #10826) Reported by: - dimas ........ - -2007-10-15 15:55 +0000 [r85557-85558] Russell Bryant <russell@digium.com> - - * pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c. - This also makes the code a bit safer by removing various - assumptions about sizes. (No vulnerabilities, though) (closes - issue #10977) Reported by: dimas Patches: dundiparser.patch - uploaded by dimas (license 88) - - * /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) | - 9 lines Ensure the buffer passed to ast_canmatch_extension() is - properly initialized so that it is null terminated. (issue - #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by - dimas (license 88) - small mods by me ........ - -2007-10-15 15:26 +0000 [r85555] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c: Allow RTP structure registration - -2007-10-15 15:07 +0000 [r85553-85554] Joshua Colp <jcolp@digium.com> - - * main/frame.c: Add packetization data for G.722. (closes issue - #10900) Reported by: andrew Patches: frame.diff uploaded by - andrew (license 240) - - * /, main/rtp.c: Merged revisions 85552 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 - lines If Monitor or a spy was added to a P2P or native bridged - channel bring the channel back to the generic bridging core so - the monitor or spy operations work. (closes issue #10943) - Reported by: julianjm ........ - -2007-10-15 13:51 +0000 [r85551] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Allocate more space for the base64 output we - need to generate. Closes issue #10913, reported by tootai, who - graciously granted us access to his Asterisk server, thanks! - Daniel, feel free to reopen the bug in case you can reproduce - this on 1.4. - -2007-10-15 13:44 +0000 [r85539-85550] Russell Bryant <russell@digium.com> - - * main/cli.c: Move the CLI commands that were in builtins[] into - the cli_cli[] array of CLI commands and remove the cli_iterator - struct. This gets tab completion working again. (closes issue - #10970) Reported by: jamesgolovich Patches: - asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license - 176) - - * doc/tex/jitterbuffer.tex, doc/tex/extensions.tex, - doc/tex/channelvariables.tex, doc/tex/ael.tex, - doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex, - doc/tex/dundi.tex, doc/tex/security.tex, - doc/tex/configuration.tex, doc/tex/ajam.tex, - doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex, - doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex, - doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c, - doc/tex/localchannel.tex, doc/tex/cdrdriver.tex, - doc/tex/queuelog.tex: Another major doc directory update from - IgorG. This patch includes - Many uses of the astlisting - environment around verbatim text to ensure that it gets properly - formatted and doesn't run off the page. - Update some things that - have been deprecated. - Add escaping as needed - and more ... - (closes issue #10978) Reported by: IgorG Patches: - texdoc-85542-1.patch uploaded by IgorG (license 20) - - * /, main/asterisk.c: Merged revisions 85545 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) | - 7 lines Make sure remote consoles unmute themselves again after - reconnecting. (closes issue #10847) Reported by: atis Patches: - console_unmute_on_reconnect.patch uploaded by atis (license 242) - ........ - - * /, main/utils.c: Merged revisions 85543 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) | - 8 lines Make sure that the base64 decoder returns a terminated - string. (closes issue #10979) Reported by: ys Patches: - util.c.diff uploaded by ys (license 281) - small mods by me - ........ - - * configure, configure.ac: Change the configure script to check for - a function that was recently added to libss7. - - * /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) | - 7 lines Don't create the context for users in users.conf until we - know at least one user exists. (closes issue #10971) Reported by: - dimas Patches: pbxconfig.patch uploaded by dimas (license 88) - ........ - - * doc/tex/backtrace.tex (added): When merging the last - documentation update, I forgot to "svn add" a file. Here it is. - (closes issue #10962) - -2007-10-13 08:38 +0000 [r85535] James Golovich <james@gnuinter.net> - - * main/cli.c: Fix compiling cli.c due to differences with new cli - system (closes issue 0010966) - -2007-10-13 05:53 +0000 [r85534] Russell Bryant <russell@digium.com> - - * include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged - revisions 85533 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | - 12 lines Fix an issue with console verbosity when running - asterisk -rx to execute a command and retrieve its output. The - issue was that there was no way for the main Asterisk process to - know that the remote console was connecting in the -rx mode. The - way that James has fixed this is to have all remote consoles - muted by default. Then, regular remote consoles automatically - execute a CLI command to unmute themselves when they first start - up. (closes issue #10847) Reported by: atis Patches: - asterisk-consolemute.diff.txt uploaded by jamesgolovich (license - 176) ........ - -2007-10-12 20:06 +0000 [r85527] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: Allow for the - position announcement to be turned off if desired. (closes issue - #8515, reported by bruno_rocha, initial patch by bruno_rocha, - final patch by qwell) - -2007-10-12 19:41 +0000 [r85525-85526] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to - finish the last of the charge_number patch up #10916 - - * channels/chan_zap.c: Add support for receive charge number in - dialplan #10916 - -2007-10-12 18:37 +0000 [r85522-85524] Tilghman Lesher <tlesher@digium.com> - - * doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions - 85523 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007) - | 2 lines Change Digium address ........ - - * funcs/func_strings.c: Enable ranges, hexadecimal, octal, and - special backslashed characters for the FILTER function - -2007-10-12 15:50 +0000 [r85516-85519] Russell Bryant <russell@digium.com> - - * doc/tex/odbcstorage.tex, doc/tex/extensions.tex, - doc/tex/channelvariables.tex, doc/tex/ael.tex, - doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex, - doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex, - doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex, - doc/tex/localchannel.tex, doc/tex/ices.tex, - doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory - improvements, including: - Added development section - (backtrace.tex) - Correct filesystem path formating - Replace all - "|" argument separator to "," - Endless count of spaces at the - end of line - Using astlisting to make listings do not take so - much place - Take back ASTRISKVERSION on first page - Make - localchannel.tex readable by inserting extra end of lines (closes - issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch - uploaded by IgorG (license 20) - - * res/res_smdi.c, /: Merged revisions 85517 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) | - 3 lines Fix a spelling error in a log message. SMDI, not SDMI. - (closes issue #10959) ........ - - * /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) | - 7 lines Fix the potential use of an uninitialized buffer in a log - message. (closes issue #10958) Reported by: dimas Patches: - realtime.patch uploaded by dimas (license 88) ........ - -2007-10-11 22:42 +0000 [r85474-85499] Matthew Fredrickson <creslin@digium.com> - - * apps/app_dial.c: Make sure we propogate ANI2 to the outbound - channel - - * funcs/func_callerid.c: See if I can fix this borked ANI2 code I - added - - * channels/chan_zap.c: Make sure we set the ANI2 field for PRI - - * funcs/func_callerid.c: Add ANI2 support to func_callerid - - * channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916 - - * channels/chan_zap.c: Add CCR test support #10916 - -2007-10-11 19:03 +0000 [r85460] Russell Bryant <russell@digium.com> - - * main/udptl.c, main/threadstorage.c, res/res_limit.c, - main/translate.c, res/res_crypto.c, res/res_convert.c, - channels/iax2-provision.c, channels/chan_gtalk.c, - channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c, - main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c, - pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c, - pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c, - main/asterisk.c, main/db.c, channels/chan_mgcp.c, - res/res_clioriginate.c: Merge a ton of NEW_CLI conversions. - Thanks to everyone that helped out! :) (closes issue #10724) - Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel - (license 64) chan_oss.c.patch uploaded by eliel (license 64) - chan_mgcp.c.patch2 uploaded by eliel (license 64) - pbx_config.c.patch uploaded by seanbright (license 71) - iax2-provision.c.patch uploaded by eliel (license 64) - chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch - uploaded by seanbright (license 71) file.c.patch uploaded by - seanbright (license 71) image.c.patch uploaded by seanbright - (license 71) cli.c.patch uploaded by moy (license 222) - astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch - uploaded by moy (license 222) res_limit.c.patch uploaded by - seanbright (license 71) res_convert.c.patch uploaded by - seanbright (license 71) res_crypto.c.patch uploaded by seanbright - (license 71) app_osplookup.c.patch uploaded by seanbright - (license 71) app_rpt.c.patch uploaded by seanbright (license 71) - app_mixmonitor.c.patch uploaded by seanbright (license 71) - channel.c.patch uploaded by seanbright (license 71) - translate.c.patch uploaded by seanbright (license 71) - udptl.c.patch uploaded by seanbright (license 71) - threadstorage.c.patch uploaded by seanbright (license 71) - db.c.patch uploaded by seanbright (license 71) cdr.c.patch - uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy - (license 222) app_osplookup-rev83558.patch uploaded by moy - (license 222) res_clioriginate.c.patch uploaded by moy (license - 222) - -2007-10-11 17:17 +0000 [r85431-85444] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Let's hard code this until I fix it - - * channels/chan_zap.c: Make sure we are clean to build without - libpri - -2007-10-11 04:40 +0000 [r85357] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 85356 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007) - | 2 lines A dollar sign by itself, not indicating a start of a - variable or expression prematurely ends substitution (closes - issue #10939) ........ - -2007-10-10 16:01 +0000 [r85317] Russell Bryant <russell@digium.com> - - * include/asterisk/file.h, /: Merged revisions 85316 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 - Oct 2007) | 6 lines I introduced a new member to the - ast_filestream struct in 1.4.12, but put it in the middle of the - struct, instead of at the end. One of the Debian folks, paravoid, - pointed out that this breaks binary compatability with modules - compiled against older headers. So, I'm moving the new member to - the end of the struct to resolve the situation. ........ - -2007-10-10 14:43 +0000 [r85281] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4 - lines If devicestate is passed a port number strip it out. - (closes issue #10930) Reported by: ibc ........ - -2007-10-10 14:38 +0000 [r85279] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct - 2007) | 5 lines A bunch of changes from sprintf to snprintf. See - security advisory AST-2002-022 ........ - -2007-10-10 14:30 +0000 [r85234-85278] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6 - lines Add support for handling a 182 Queued response. (closes - issue #10924) Reported by: ramonpeek Patches: queued-182.diff - uploaded by ramonpeek (license 266) ........ - - * /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6 - lines Close voicemail message description file if duration did - not meet the minimum, or else we will eventually run out of file - descriptors. (closes issue #10918) Reported by: brak2718 Patches: - vm1.4.12.1.patch uploaded by brak2718 (license 279) ........ - - * main/logger.c: Process outstanding log messages before shutting - down the logger thread. (closes issue #10933) Reported by: - sperreault - -2007-10-10 06:48 +0000 [r85197] Luigi Rizzo <rizzo@icir.org> - - * bootstrap.sh: Adapt the autotools names to different versions of - FreeBSD (and open the way to better adaptation for other - platforms as well). - -2007-10-10 06:41 +0000 [r85196] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10 - Oct 2007) | 2 lines use a macro instead of an inline function, so - that backtraces will report the caller of ast_frame_free() - properly ........ - -2007-10-09 22:35 +0000 [r85177] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Patch to add one-touch parking for queues. - (closes issue #10869, reported and patched by bluecrow76) - -2007-10-09 22:21 +0000 [r85140-85176] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged - revisions 85158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007) - | 5 lines This commit fixes the following issues: - Deadlock in - ast_write (issue #10406) - Deadlock in ast_read (issue #10406) - - Possible mutex initialization error in lock.h (issue #10571) - ........ - - * apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c, - apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c, - utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove - redundant includes (patch by snuffy) (Closes issue #10922) - -2007-10-09 15:12 +0000 [r85097-85098] Russell Bryant <russell@digium.com> - - * CHANGES: Note jitterbuffer support for chan_local in CHANGES - - * channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer - support for chan_local. To enable it, you use the 'j' option in - the Dial command. The 'j' option _must_ be used in conjunction - with the 'n' option. This feature will allow you to use the - existing jitterbuffer implementation to put a jitterbuffer on - incoming SIP calls connecting to Asterisk applications by putting - a local channel in the middle. - -2007-10-09 14:31 +0000 [r84991-85094] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4 - lines Don't perform a reinvite if a transfer is in progress. - (issue #10915) Reported by: ramonpeek ........ - - * /, main/rtp.c: Merged revisions 85057 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4 - lines Only update codec information if the channel has a - technology private structure. (issue #10915) Reported by: - ramonpeek ........ - - * res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c, - main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it - compiles when MTX Profiling is enabled. (closes issue #10898) - Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by - qwell (license 4) - - * /, main/rtp.c: Merged revisions 85023 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4 - lines Update codec information as well as address when doing hold - reinvites. (issue #10868) Reported by: mavince ........ - - * main/channel.c, /: Merged revisions 84990 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4 - lines Don't keep trying to native bridge if either of the - channels are involved in a masquerade operation to be done. - (closes issue #10696) Reported by: tbelder ........ - -2007-10-08 03:29 +0000 [r84958] Russell Bryant <russell@digium.com> - - * /, Makefile.rules: Merged revisions 84957 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) | - 6 lines Enable file dependency tracking for _all_ builds, and not - just for builds with dev-mode enabled. I have seen enough - problems caused by this that I don't think it's worth keeping. I - want to continue to encourage anybody that is interested to - continue to run Asterisk from svn. Furthermore, I do not want - their systems to break when we change a structure definition in a - header file. :) ........ - -2007-10-07 16:28 +0000 [r84891-84939] Philippe Sultan <philippe.sultan@gmail.com> - - * configs/jabber.conf.sample, include/asterisk/jabber.h, - res/res_jabber.c: Make the status and priority configurable. - Closes issue #10785, patch by Luke-Jr, thanks! - - * /, res/res_jabber.c: Merged revisions 84902 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) - | 5 lines Presence packets from a client who's connected with our - Jabber ID are valid, therefore, those clients must be considered - as buddies. The resource string helps us make the distinction - between clients. Closes issue #10707, reported by yusufmotiwala. - ........ - - * res/res_jabber.c: Fix indentation - - * /, res/res_jabber.c: Merged revisions 84890 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) - | 5 lines Prevent Asterisk from crashing when receiving a - presence packet without resource from a buddy that is known to - have a resource list. Revert a change I previously made, where - Asterisk could point to a freed memory location. ........ - -2007-10-05 19:48 +0000 [r84852] Tilghman Lesher <tlesher@digium.com> - - * /, main/db.c: Merged revisions 84851 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007) - | 2 lines Log exactly why we can't open the database, if we fail - (closes issue #10887) ........ - -2007-10-05 18:57 +0000 [r84819] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 84818 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 - lines Update the remembered RTP peer information when putting an - endpoint on hold or taking it off hold so that the RTP stack does - not initiate a needless reinvite. (closes issue #10868) Reported - by: mavince ........ - -2007-10-05 16:49 +0000 [r84743-84784] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) | - 4 lines Do deadlock avoidance in a couple more places. You can't - lock two channels at the same time without doing extra work to - make sure it succeeds. (closes issue #10895, patch by me) - ........ - - * main/manager.c, /: Merged revisions 84742 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) | - 3 lines Fix a copy/paste error in the description of UpdateConfig - that was pointed out by JerJer on #asterisk-dev ........ - -2007-10-04 22:58 +0000 [r84693-84726] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix - some bad logic in the counting of statistics for QueueSummary - manager event. Variables were not being reset for each additional - queue, so cumulative totals were reported on each successive - queue. 2) Add a longest hold time stat to QueueSummary manager - event. - - * /, apps/app_queue.c: Merged revisions 84692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct - 2007) | 5 lines Don't allocate space for queue members unless - it's needed. You end up deleting dynamic members on a reload. Not - good. closes issue (#10879, reported by dazza76, patched by me) - ........ - -2007-10-04 21:38 +0000 [r84691] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007) - | 2 lines callers of sig2str already add the word 'signalling' in - the appropriate place, so don't duplicate it ........ - -2007-10-04 16:56 +0000 [r84671] Tilghman Lesher <tlesher@digium.com> - - * res/res_jabber.c: Update to current coding standards, also - changing the argument delimiter to ',' (Closes issue #10876) - -2007-10-04 14:54 +0000 [r84613-84638] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 84637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4 - lines Create a duplicate of the channel's member name as the tab - completion stuff will free it. (closes issue #10884) Reported by: - adamg ........ - - * main/pbx.c: Don't register the exception function with module - information. Since it is in the core there is none and it will - explode. - -2007-10-03 23:05 +0000 [r84580-84582] Tilghman Lesher <tlesher@digium.com> - - * /, main/rtp.c: Merged revisions 84581 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007) - | 2 lines When an RFC 2833 event is sent that we don't recognize, - ignore it, don't queue a NULL digit (closes issue #10877) - ........ - - * main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h: - Create a universal exception handling extension, "e" (closes - issue #9785) - -2007-10-03 18:23 +0000 [r84512-84545] Steve Murphy <murf@digium.com> - - * /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the - CUT in AEL - - * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /, - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, - pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1 - line closes issue #10834 ; where a null input to a switch - statement results in a hangup; since switch is implemented with - extensions, and the default case is implemented with a '.', and - the '.' matches 1 or more remaining characters, the case where 0 - characters exist isn't matched, and the extension isn't matched, - and the goto fails, and a hangup occurs. Now, when a default case - is generated, it also generates a single fixed extension that - will match a null input. That extension just does a goto to the - default extension for that switch. I played with an alternate - solution, where I just tack an extra char onto all the patterns - and the goto, but not the default case's pattern. Then even a - null input will still have at least one char in it. But it made - me nervous, having that extra char in , even if that's a pretty - secret and low-level issue. ........ - -2007-10-02 20:07 +0000 [r84475] Russell Bryant <russell@digium.com> - - * Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) | - 5 lines * Don't build the menuselect-tree for the tarball, as it - requires running the configure script first * Change the Makefile - to note that menuselect-tree depends on the configure script. - ........ - -2007-10-02 19:02 +0000 [r84432-84440] Jason Parker <jparker@digium.com> - - * /, res/res_features.c: Merged revisions 84410 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500 - (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel - before return on failure. Issue 10821, patch by Ivan ........ - -2007-10-02 18:12 +0000 [r84405] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Add MSet for people who prefer the old, deprecated - syntax of Set (Closes issue #10549) - -2007-10-02 14:13 +0000 [r84371] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | - 6 lines Use snprintf instead of sprintf in one place. There is no - vulnerability here due to various buffer sizes around the code, - but I still didn't like seeing a non length-limited copy of data - coming off of the wire into a stack buffer, as this would be a - problem in the future if buffer sizes elsewhere got changed or - size limitations removed ... ........ - -2007-10-02 13:58 +0000 [r84368] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Don't swap channel priority if using epoll as polling - should/will only happen off the first channel. (closes issue - #10867) Reported by: phsultan - -2007-10-01 23:33 +0000 [r84327-84331] Steve Murphy <murf@digium.com> - - * utils/check_expr.c: OK. THis a DEBUG_THREADS situation. - - * utils/check_expr.c: picky gcc versions... sigh. - - * utils/check_expr.c: This mod will allow check_expr to compile in - the presence of DEBUG_THREAD situations. At least, it does for - me. And it's less expensive than several other approaches I - tried. - - * res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y, - pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1 - line closes issue #10777 -- by returning a null for the parse - tree when there's really nothing there, and making sure we don't - try to do checking on a null tree. ........ - -2007-10-01 21:54 +0000 [r84300] Jason Parker <jparker@digium.com> - - * Makefile, /, Makefile.rules, channels/Makefile: Merged revisions - 84291 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 - lines Add dist-clean support for subdirs. Change h323 to only - remove the Makefile on a dist-clean, rather than a clean. This - fixes a bug I found with trying to run make after a make clean - ........ - -2007-10-01 21:31 +0000 [r84275] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged - revisions 84274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007) - | 1 line moved get_base_channel() code from action_redirect to - ast_channel_masquerade() for issue 7706 and BE-160 ........ - -2007-10-01 21:15 +0000 [r84207-84272] Russell Bryant <russell@digium.com> - - * /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) | - 4 lines Fulfull a feature request from Qwell on the "core show - locks" output. It will now note the lock type for each lock that - a thread holds. (mutex, rdlock, or wrlock) ........ - - * /, res/res_agi.c: Merged revisions 84236 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) | - 5 lines Add another sanity check in the AGI read loop. We really - don't care about EAGAIN unless we didn't read an entire line. If - there is a newline at the end if the read buffer, break, because - we got the whole thing. (reported and patched by bmd) ........ - - * /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01 - Oct 2007) | 2 lines Show rwlocks in the "core show locks" output. - Before, it only showed mutexes. ........ - -2007-10-01 15:57 +0000 [r84176] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Check to make sure a structure pointer is - non-NULL before touching it... crashing is bad, mmmk? (closes - issue #10831) Reported by: eliel Patches: chan_sip.c.patch - uploaded by eliel (license 64) - -2007-10-01 15:34 +0000 [r84167-84174] Russell Bryant <russell@digium.com> - - * main/say.c: Change simple uses of snprintf to ast_copy_string. - This was provided by mvanbaak as a part of issue #10843, but this - part didn't apply because of a patch I applied right beforehand. - - * channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c, - apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, - main/say.c, apps/app_minivm.c, pbx/dundi-parser.c, - channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c, - main/rtp.c, channels/chan_mgcp.c: Corydon posted this janitor - project to the bug tracker and mvanbaak provided a patch for it. - It replaces a bunch of simple calls to snprintf with - ast_copy_string (closes issue #10843) Reported by: Corydon76 - Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) - - * main/say.c: Simplify code by using the -= and %= operators. - (closes issue #10848) Reported by: opticron Patches: saymod.diff - uploaded by opticron (license 267) - - * codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The - trunk version of this patch also includes a couple more small - clean fixes from IgorG. Merged revisions 84170 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) | - 3 lines Remove another file in "make clean". (closes issue - #10814, paravoid) ........ - - * main/cli.c: Don't set the full command string until after - verifying that there is not another CLI command with the same - command text registered. This prevents a crash if someone - accidentally calls ast_cli_register() on the same CLI command - data twice. This also fixes a small bug where the helpers list - would get unlocked without being locked if building the full - command failed. (closes issue #10858, reported by jamesgolovich, - patched by me) - - * configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new - option for files-based music on hold to ensure that the sort - order of the files is alphabetical. (closes issue #10855) - Reported by: jamesgolovich Patches: - asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license - 176) - - * apps/app_dial.c, /: Merged revisions 84166 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | - 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ - -2007-10-01 14:21 +0000 [r84159-84165] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Add MP4 to part of the SDP code. (closes - issue #10820) Reported by: ruikubo Patches: chan_sip.patch - uploaded by ruikubo (license 250) - - * main/dnsmgr.c: Don't register the dnsmgr refresh CLI command - twice. (closes issue #10856) Reported by: jamesgolovich Patches: - asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license - 176) - - * /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct - 2007) | 6 lines Fix randomness. save_pos was being set to 0 - initially instead of -1, causing it to jump to position 0 when - moh started. (closes issue #10859) Reported by: jamesgolovich - Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich - (license 176) ........ - - * apps/app_dial.c, /: Merged revisions 84158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 - lines Only attempt early bridging if the options given to Dial() - permit it. (closes issue #10861) Reported by: peekyb ........ - -2007-09-30 20:06 +0000 [r84143-84147] Russell Bryant <russell@digium.com> - - * /, include/asterisk/module.h: Merged revisions 84146 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30 - Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++ - modules. The load and reload parameters were in the wrong place. - (closes issue #10846, alebm) ........ - - * funcs/func_lock.c: * The documentation for the LOCK() function - says that it will block for up to 3 seconds while waiting on a - lock when other locks are currently held to avoid deadlocks. - Change the code to reflect this. * Since trying to grab a lock - may block for some time, put the channel in autoservice so that - audio is still read from the channel and that any active - generators on the channel don't pause. - -2007-09-29 23:47 +0000 [r84134-84137] Steve Murphy <murf@digium.com> - - * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1 - line This issue sort of closes 10786; All config files support - #include with globbing (you know, *,[chars],?,{list,list},etc), - so I've updated the AEL system to support this also. ........ - - * pbx/ael/ael-test/ael-ntest22/t2 (added), - pbx/ael/ael-test/ael-ntest22/t3 (added), - pbx/ael/ael-test/ael-ntest22/extensions.ael (added), - pbx/ael/ael-test/ael-ntest22 (added), - pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), - pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), - pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), - pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 - (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), - pbx/ael/ael-test/ref.ael-test3, - pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), - pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), - pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), - pbx/ael/ael-test/ael-ntest22/qq.ael (added), - pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL - affected a small number of tests. Added a regression test for - glob'd includes - -2007-09-29 18:21 +0000 [r84130] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_manager.c: Set enablecdr at the end of re-reading the - config file (Closes issue #10852) - -2007-09-29 00:19 +0000 [r84115] Matthew Fredrickson <creslin@digium.com> - - * main/translate.c: Let's use process time instead of wall clock - time for show translation - -2007-09-28 14:35 +0000 [r84050-84080] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: Autoconf requires version 2.60, not - 2.59, to process (Closes issue #10842) - - * /, main/say.c: Merged revisions 84078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007) - | 2 lines Correct pronunciations of numbers for .nl (Closes issue - #10837) ........ - - * main/channel.c, /: Merged revisions 84049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007) - | 3 lines Avoid a deadlock with ALL of the locks in the - masquerade function, not just the pairs of channels. (Closes - issue #10406) ........ - -2007-09-27 23:18 +0000 [r84019] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/manager.c, /, channels/chan_agent.c, - include/asterisk/channel.h: Merged revisions 84018 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27 - Sep 2007) | 1 line if an Agent is redirected, the base channel - should actually be redirected. This was causing multiple issues, - especially issue 7706 and BE-160 ........ - -2007-09-27 00:08 +0000 [r83978-83986] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007) - | 2 lines avoid the weird usage of assert() in the ALSA header - files that gcc 4.2 wants to complain about ........ - - * res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const - pointer warnings - -2007-09-27 00:02 +0000 [r83911-83977] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) | - 1 line remove a todo item that has been completed ........ - - * /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) | - 2 lines I changed my mind ... I think this should be a - LOG_NOTICE. ........ - - * /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | - 5 lines Add a log message that was requested by the masses in the - developer tutorial session at Astricon. chan_sip did not output - any message when a call was rejected because the extension was - not found. This adds a verbose message (at verbose level 3) to - note when this happens. ........ - - * /: Merged revisions 83910 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) | - 3 lines Fix building chan_misdn under dev-mode. (please run the - configure script with --enable-dev-mode so this doesn't happen - again ...) ........ - -2007-09-26 18:43 +0000 [r83880] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007) - | 2 lines Remove unused 4k of memory on the program stack (closes - issue #10827) ........ - -2007-09-26 06:53 +0000 [r83849-83864] Russell Bryant <russell@digium.com> - - * include/asterisk/event.h: fix a typo in a comment - - * include/asterisk/file.h: Change function documentation to use - doxygen tags. (Really, I just needed to make some minor change in - trunk to test something with automerge ...) - -2007-09-25 23:14 +0000 [r83834] Matthew Fredrickson <creslin@digium.com> - - * doc/ss7.txt: Fix typo in readme - -2007-09-25 21:06 +0000 [r83819] Russell Bryant <russell@digium.com> - - * include/asterisk/devicestate.h: Don't note that functions are - deprecated in favor of themselves. This was found by showing a - very poor example doxygen function in a presentation this - morning. :) - -2007-09-25 16:34 +0000 [r83804] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Added a CLI command that shows our buddy list, - as suggested by Daniel McKeehan, thanks! - -2007-09-25 14:18 +0000 [r83774] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: Merged revisions 83773 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007) - | 2 lines jmls pointed out that unsetting the group and setting - the group to the blank string aren't quite the same. ........ - -2007-09-25 13:41 +0000 [r83758] Joshua Colp <jcolp@digium.com> - - * res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a - value without freeing the previous result is bad, mmmk? - -2007-09-25 09:07 +0000 [r83743] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c, include/asterisk/jingle.h: Comply with - latest XEP-0166, XEP-0167, XEP-0176. No real Jingle - implementation being available, testing was made using two - Asterisk servers relaying SIP calls over their Jingle channels: - SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- - [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was - possible to test the code in both ways, and make the Jingle - channel comply with the latest specifications. No sound available - yet. Main modifications include : - modified the - 'jingle_candidate' structure and the 'jingle_create_candidates' - function according to XEP-0176 ; - modified the 'jingle_action' - function in order to properly terminate a Jingle session, in - conformance with XEP-0166 ; - modified username format used in - STUN requests ; - actually make the bindaddr configuration field - useable. Todo : - set audio paths up (no native bridging) ; - - make the CLI gtalk functions available to jingle ; - clean up the - storage space used in strings. - -2007-09-25 08:09 +0000 [r83741] Russell Bryant <russell@digium.com> - - * utils/Makefile, utils: Add some files to the utils directory - svn:ignore and Makefile clean target (closes issue #10808, - reported by mvanbaak) - -2007-09-24 22:06 +0000 [r83696-83726] Tilghman Lesher <tlesher@digium.com> - - * Makefile, main/asterisk.c: Permit custom locations for astdb and - the keys directory (though default to the current locations) - (Closes issue #10267) - - * /, build_tools/make_defaults_h: Merged revisions 83695 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007) - | 4 lines In the source, keys are relative to the datadir, not - varlib (which is the same in most cases, but it's good to be - accurate). Closes issue #10811 ........ - -2007-09-24 17:10 +0000 [r83671] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's - patch for configurable SDP origin-field username and session - field, closes issue# 10795 - -2007-09-24 17:00 +0000 [r83656] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: interface_exists_global was never returning 1. - Most likely an error from my merge on Friday. (closes issue - #10817, reported and patched by snar, patch simplified by me) - -2007-09-24 16:42 +0000 [r83654-83655] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: Merged revisions 83637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007) - | 3 lines Making change to group splitting, as discussed on the - -dev list. The main effect of this will be to permit - Set(GROUP([cat])=), i.e. unsetting a group. ........ - -2007-09-22 19:54 +0000 [r83575-83590] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, /: Merged revisions 83589 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1 - line This closes issue #10788 -- The exact same fixes are made - here for the first arg in the for(arg1; arg2; arg3) {} statement, - as were done for the 3rd arg. It can now be an assignment that - will embedded in a Set() app, or a macro call, or an app call. - ........ - - * res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1 - line This closes issue #10788 -- the 3rd arg in the for statement - is now wrapped in Set() only if there's an '=' in that string. - Otherwise, if it begins with '&', then a Macro call is generated; - otherwise it is made into an app call. A bit more accomodating, - keeps the new guys happy, and the guys with ael-1 code should be - happy, too ........ - -2007-09-22 17:37 +0000 [r83574] Matthew Fredrickson <creslin@digium.com> - - * doc/ss7.txt: Fix potential point of confusion - -2007-09-22 14:45 +0000 [r83517-83545] Tilghman Lesher <tlesher@digium.com> - - * utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix - build of check_expr and hashtest2 when DEBUG_THREADLOCAL is - defined - - * main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload - manager commands, which supplement the need to have Command - privilege. (closes issue #10736) - - * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, - main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for - FreeBSD... testing for every conceivable math function now - -2007-09-21 19:55 +0000 [r83500] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix compilation errors in CLI command - updates to SS7 CLI commands - -2007-09-21 19:54 +0000 [r83499] Matthew Fredrickson <creslin@digium.com> - - * doc/ss7.txt (added): Add an SS7 readme for setup and use of - libss7 and asterisk - -2007-09-21 18:41 +0000 [r83484] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c: Fix some areas where we were still using '|' - for an argument delimiter (closes issue #10793) - -2007-09-21 18:27 +0000 [r83483] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Update app_queue to use commas as application - argument separators. (closes issue #10793, snar) - -2007-09-21 17:36 +0000 [r83466] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_manager.c: Fix cdr_manager, such that if the config file - is created past load, it'll start logging (and conversely, if the - config file is destroyed or deactivated, the logging is - disabled). Reported by Juggie via IRC, fix by me. - -2007-09-21 14:40 +0000 [r83433] Russell Bryant <russell@digium.com> - - * res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c, - main/db1-ast/hash/hash.c, include/asterisk/channel.h, - channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c, - main/cdr.c, main/channel.c, channels/chan_misdn.c, - main/ast_expr2f.c, main/file.c, include/asterisk/sched.h, - channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c, - main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged - revisions 83432 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | - 4 lines gcc 4.2 has a new set of warnings dealing with cosnt - pointers. This set of changes gets all of Asterisk (minus - chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, - patch from qwell) ........ - -2007-09-21 14:25 +0000 [r83431] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, - main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the - presence of trunc and round, and make the ISOC99 detection a - little more sane (closes issue #10776) - -2007-09-20 23:14 +0000 [r83381] Jason Parker <jparker@digium.com> - - * apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More - NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch - uploaded by moy (license 222) app_minivm.c.patch uploaded by - eliel (license 64) astmm.c.patch uploaded by eliel (license 64) - -2007-09-20 21:37 +0000 [r83350-83351] Mark Michelson <mmichelson@digium.com> - - * /: Oops. Getting rid of svnmerge-integrated and automerge stuff - - * /, apps/app_queue.c: Merging changes from queue_refcount_trunk - into trunk. Refcounted queues now in place. - -2007-09-20 21:17 +0000 [r83293-83349] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: Merged revisions 83348 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) | - 4 lines When daemonizing, don't change working directory to "/". - It makes it not be able to do a core dump when not running as - uid=root. (closes issue #10766, xrg) ........ - - * /, contrib/scripts/safe_asterisk: Merged revisions 83316 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) | - 3 lines Change safe_asterisk to explicitly ask for /bin/bash, as - it uses bashisms. (closes issue #10772, reported by culrich) - ........ - - * main/dsp.c: trivial formatting change - - * main/asterisk.c: trivial formatting change - - * main/app.c: minor spelling fixes in a comment - - * main/app.c: minor grammar fix - - * channels/chan_sip.c: fix spelling in a comment - - * main/asterisk.c: trivial formatting change - -2007-09-20 19:05 +0000 [r83251-83278] Jason Parker <jparker@digium.com> - - * doc/modules.txt: Fix a trivial typo, to test our new commit bot - - * /, apps/app_disa.c: Merged revisions 83246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8 - lines If # is pressed after dialing an extension in DISA, stop - trying to collect more digits. (closes issue #10754) Reported by: - atis Patches: app_disa.c.branch.patch uploaded by atis (license - 242) app_disa.c.trunk.patch uploaded by atis (license 242) - ........ - -2007-09-20 16:28 +0000 [r83234] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7 - lines Make sure the minimum T1 timer value is obeyed in all - cases. (closes issue #10768) Reported by: flefoll Patches: - chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license - 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll - (license 244) ........ - -2007-09-20 16:27 +0000 [r83233] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Don't start the event processing thread until - after forking. (reported by Simon on the -dev list, thanks!) - -2007-09-20 16:19 +0000 [r83229-83231] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7 - lines Fix a minor spelling error. (closes issue #10769) Reported - by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded - by flefoll (license 244) chan_sip.c.br14.83070.inita-patch - uploaded by flefoll (license 244) ........ - - * pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks - in pbx_dundi, cdr_pgsql, and the configuration file parser. - -2007-09-19 23:16 +0000 [r83213] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c, - apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724) - Patches: chan_zap.c.patch uploaded by moy (license 222) - app_queue.c.patch uploaded by eliel (license 64) - app_voicemail.c.patch uploaded by eliel (license 64) - app_meetme.c.patch uploaded by eliel (license 64) - -2007-09-19 20:06 +0000 [r83182-83183] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme - for me today?) - - * res/res_adsi.c: Clean up code in res_adsi. - -2007-09-19 19:54 +0000 [r83176-83181] Russell Bryant <russell@digium.com> - - * funcs/func_shell.c: put the channel in autoservice when executing - func_shell - - * /, apps/app_system.c: Merged revisions 83179 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) | - 5 lines The System() and TrySystem() applications can take a - substantial amount of time to execute while not servicing the - channel. So, put the channel in autoservice while the command is - being executed. (closes issue #10726, reported by mnicholson) - ........ - - * funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) | - 4 lines Using curl can take a substantial amount of time, so the - channel should be autoserviced while waiting for it to complete. - (closes issue #10725, reported by mnicholson) ........ - - * /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) | - 8 lines When handling a reload of chan_iax2, don't use an - ao2_callback() to POKE all peers. Instead, use an iterator. By - using an iterator, the peers container is not locked while the - POKE is being done. It can cause a deadlock if the peers - container is locked because poking a peer will try to lock pvt - structs, while there is a lot of other code that will hold a pvt - lock when trying to go lock the peers container. (reported to me - directly by Loic Didelot. Thank you for the debug info!) ........ - -2007-09-19 17:22 +0000 [r83155-83157] Joshua Colp <jcolp@digium.com> - - * apps/app_db.c: Fix indentation in app_db. - - * apps/app_authenticate.c: Clean up code in app_authenticate. - - * apps/app_adsiprog.c: Clean up code in app_adsiprog. - -2007-09-19 15:11 +0000 [r83126] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 83121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) | - 4 lines Fix up another potential race condition. Do the loop - decrementing use count on events with the eventq protected from - being changed. (reported on IRC by Ivan) ........ - -2007-09-19 15:08 +0000 [r83105-83114] Joshua Colp <jcolp@digium.com> - - * apps/app_disa.c: DISA only needs to know about the end of DTMF, - not the beginning/duration. - - * apps/app_disa.c: Clean up app_disa code a bit. - -2007-09-19 13:55 +0000 [r83076] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c: Replace Google namespace occurrences with - Jingle. The former namespace is handled by chan_gtalk. - -2007-09-19 13:49 +0000 [r83073-83075] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 83074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6 - lines Protect the CDR record from modification by pbx_exec so - that the application data contains the Queue data. (closes issue - #10761) Reported by: snar Patches: app-queue-mixmonitor.patch - uploaded by snar (license 245) ........ - - * main/manager.c: Extend manager show connected with additional - information. (closes issue #10757) Reported by: outtolunc - Patches: manager.c.sessionstart.diff uploaded by outtolunc - (license 237) - -2007-09-19 13:29 +0000 [r83072] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c: Remove namespaces in payload-type tags. - -2007-09-19 13:21 +0000 [r83071] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6 - lines (closes issue #10760) Reported by: dimas Patches: - chan_sip.patch uploaded by dimas (license 88) Read in - subscribecontext option in general to be the default. ........ - -2007-09-19 12:23 +0000 [r83055] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c, include/asterisk/jingle.h: Transmit - proper invitation, thus conforming to XEP-0166 (Jingle general - specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 - (Jingle ICE Transport). - -2007-09-19 09:48 +0000 [r83025] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, - channels/misdn_config.c: Merged revisions 83023-83024 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) | - 1 line added 'astdtmf' option to allow configuring the asterisk - dtmf detector instead of the mISDN_dsp ones. also added the patch - from irroot #10190, so that dtmf tones detected by the asterisk - detector are passed outofband to asterisk, to make any use of - dtmf tones at all. ........ r83024 | crichter | 2007-09-19 - 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which - violates the coding guidelines. ........ - -2007-09-19 00:21 +0000 [r82993] Russell Bryant <russell@digium.com> - - * /, apps/app_flash.c: Merged revisions 82992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) | - 4 lines Change the description of app_flash to note how it can be - a useful tool instead of just saying that it is generally a - worthless feature. (Thanks to Jim Van Meggelen for pointing it - out and providing the proposed text) ........ - -2007-09-18 23:42 +0000 [r82962] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 82961 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2 - lines Initialize a variable to NULL to make the world happy. - ........ - -2007-09-18 22:46 +0000 [r82931] Russell Bryant <russell@digium.com> - - * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | - 11 lines Add a new patch to handle interrupting the fgets() call - when using FastAGI. This version of the patch maintains the - original behavior of the code when not using FastAGI. (closes - issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch - uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch - uploaded by juggie (license 24) Slight mods by me Tested by: - juggie, festr ........ - -2007-09-18 22:43 +0000 [r82871-82930] Jason Parker <jparker@digium.com> - - * main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c, - channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, - res/res_musiconhold.c, res/res_jabber.c, main/manager.c, - res/res_agi.c, channels/chan_features.c, main/logger.c, - main/http.c, channels/chan_alsa.c, res/res_realtime.c, - res/res_odbc.c: (issue #10724) Reported by: eliel Patches: - res_features.c.patch uploaded by eliel (license 64) - res_agi.c.patch uploaded by seanbright (license 71) - res_musiconhold.c.patch uploaded by seanbright (license 71) - pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded - by moy (license 222) frame.c.patch uploaded by moy (license 222) - manager.c.patch uploaded by moy (license 222) http.c.patch - uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy - (license 222) res_realtime.c.patch uploaded by eliel (license 64) - res_odbc.c.patch uploaded by seanbright (license 71) - res_jabber.c.patch uploaded by eliel (license 64) - chan_local.c.patch uploaded by eliel (license 64) - chan_agent.c.patch uploaded by eliel (license 64) - chan_alsa.c.patch uploaded by eliel (license 64) - chan_features.c.patch uploaded by eliel (license 64) - chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch - (includes all of the above patches) uploaded by seanbright - (license 71) Convert many CLI commands to the NEW_CLI format. - - * configs/voicemail.conf.sample, apps/app_voicemail.c: (closes - issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff - uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by - qwell (license 4) Tested by: callguy, ruffle Add an option to - disable the automatic moving of "heard" messages to the Old - folder. - -2007-09-18 20:59 +0000 [r82868] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 82867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) | - 10 lines Fix a memory leak that can occur on systems under higher - load. The issue is that when events are appended to the master - event queue, they use the number of active sessions as a use - count so it will know when all active sessions at the time the - event happened have consumed it. However, the handling of the - number of sessions was not properly synchronized, so the use - count was not always correct, causing an event to disappear - early, or get stuck in the event queue for forever. (closes issue - #9238, reported by bweschke, patch from Ivan, modified by me) - ........ - -2007-09-18 20:10 +0000 [r82866] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 82865 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep - 2007) | 4 lines Moving the logic for handling an empty membername - to the create_member function so that there is a common place - where this occurs instead of being spread out to several - different places. ........ - -2007-09-18 19:06 +0000 [r82835] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_queue.c: Merged revisions 82834 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007) - | 2 lines there is no need for conditional logic to select - ->interface or ->membername, snince ->membername will always be - populated ........ - -2007-09-18 16:34 +0000 [r82803] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) | - 4 lines When copying the contents from the wildcard peer, do a - deep copy instead of shallow copy so that it doesn't crash when - beging destroyed. (closes issue #10546, patch by me) ........ - -2007-09-18 16:16 +0000 [r82800] Jason Parker <jparker@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: (closes issue - #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch - uploaded by snar (license 245) queues.conf.patch uploaded by snar - (license 245) Add an updatecdr option to queues.conf, so that if - a "member name" is specified, the cdr record will be updated with - that, rather than the channel. - -2007-09-18 16:14 +0000 [r82776-82793] Russell Bryant <russell@digium.com> - - * include/asterisk/threadstorage.h: Make sure that libpthread - doesn't try to call free() directly when MALLOC_DEBUG is enabled. - If it does, Asterisk will crash as the address isn't the real - beginning of the allocation. - - * channels/chan_zap.c: Don't use ast_channel_lock_both() here, it - only exists in one of my branches. This is theoretically a - potential deadlock, but it's the way it was before so I'm going - to leave it this way for now. - -2007-09-18 15:29 +0000 [r82752] Jason Parker <jparker@digium.com> - - * /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500 - (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains - option in SIP sample config. Issue 10753 ........ - -2007-09-17 22:59 +0000 [r82728] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c, - channels/chan_agent.c, channels/chan_alsa.c, - channels/chan_iax2.c, channels/chan_mgcp.c: convert various - places that access the channel lock directly to use the channel - lock wrappers - -2007-09-17 21:52 +0000 [r82710-82712] Jason Parker <jparker@digium.com> - - * cdr/cdr_sqlite3_custom.c: Don't try to continue loading - cdr_sqlite3_custom on a module load failure (such as the config - not existing) Closes issue #10749, patch by seanbright. - - * configs/http.conf.sample: Fix the sample redirect to point to a - valid file in the Asterisk GUI. Closes issue #10748, patch by - bkruse - -2007-09-17 20:24 +0000 [r82595-82679] Russell Bryant <russell@digium.com> - - * doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support - for #include, var_metric, and cat_metric in res_config_sqlite - (closes issue #10738, rbraun_proformatique) - - * /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged - revisions 82676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) | - 4 lines Put a memset in ast_localtime() instead of a couple - places in app_voicemail to prevent the problem everywhere instead - of just a couple of places. (related to issue #10746) ........ - - * /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) | - 6 lines Initialize some memory to fix crashes when leaving - voicemail. This problem was fixed by running Asterisk under - valgrind. (closes issue #10746, reported by arcivanov, patched by - me) *** IMPORTANT NOTE: We need to check to see if this same bug - exists elsewhere. ........ - - * apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h, - apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, - res/res_features.c, apps/app_queue.c, channels/chan_iax2.c, - pbx/pbx_config.c: Make the MALLOC_DEBUG output for free() useful - again. After changing calls to free to be ast_free, astmm said - all calls to free were coming from utils.h - - * /, res/res_features.c: Merged revisions 82594 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) | - 5 lines Handle the case where there are multiple dynamic features - with the same digit mapping, but won't always match the activated - on/by access controls. In that case, the code needs to keep - trying features for a match. (reported by Atis on the - asterisk-dev list, patched by me) ........ - -2007-09-17 16:44 +0000 [r82593] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17 - Sep 2007) | 2 lines fix a couple of places where a logical member - name (if specified) was not used, but instead the direct - interface was listed ........ r82592 | kpfleming | 2007-09-17 - 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that - wasn't supposed to be committed... doh! ........ - -2007-09-17 14:58 +0000 [r82568] Doug Bailey <dbailey@digium.com> - - * main/http.c: Fix memory leak introduced when POST support was - added. - -2007-09-17 02:20 +0000 [r82516-82546] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10715) Reported by: - the-chopper Don't bother hanging up the new channel if it does - not exist yet. - - * main/pbx.c, /: Merged revisions 82514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4 - lines (closes issue #10734) Reported by: asgaroth Instead of - passing a NULL pointer into snprintf pass "". It makes Solaris - much happier. ........ - -2007-09-16 15:32 +0000 [r82496] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Option maxmessage should be maxsecs - per-folder, too (closes issue #10729) - -2007-09-14 21:30 +0000 [r82457] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 82444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1 - line closes issue #10668; thanks to arkadia for his patch; had to - leave out the bit about ending the previous cdr in the fork; it - would destroy current implementations. ........ - -2007-09-14 21:21 +0000 [r82454] Russell Bryant <russell@digium.com> - - * /, configs/zapata.conf.sample: Merged revisions 82435 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) | - 3 lines Add a note to help clarify the value set with the - echocancel option. (inspired by Malcolm's blog post on - blogs.digium.com about HPEC) ........ - -2007-09-14 19:49 +0000 [r82401] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Add support - in chan_skinny for sending RTP directly to the endpoints. Closes - issue #9154, patch by DEA - -2007-09-14 18:37 +0000 [r82397-82400] Mark Michelson <mmichelson@digium.com> - - * /: Blocking revision 82398 - - * /, apps/app_queue.c: Merged revisions 82396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep - 2007) | 5 lines Adding member name field to manager events where - they were missing before (closes issue #10721, reported by snar) - ........ - -2007-09-14 17:51 +0000 [r82395] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5 - lines If a channel does not have an owner, do not try to set a - channel variable. This will end up making the channel variable - global, which is not right. Closes issue #10720, patch by - flefoll. ........ - -2007-09-14 17:29 +0000 [r82393] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute - method to res_odbc (closes issue #10722) - -2007-09-14 16:02 +0000 [r82386-82391] Russell Bryant <russell@digium.com> - - * channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed), - channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c, - channels/xpmr/xpmr_coef.h: use the standard license header for - the xpmr files - - * channels/chan_usbradio.c (added), channels/xpmr (added): Add - chan_usbradio to trunk - - * /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: - Merged revisions 82385 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) | - 3 lines Add checking for libusb here, so nobody has to deal with - conflicts in the chan_usbradio-1.4 branch every time the - configure script gets changed ........ - -2007-09-14 14:44 +0000 [r82377] Mark Michelson <mmichelson@digium.com> - - * doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14 - Sep 2007) | 5 lines Fixing a typo in the coding guidelines - (closes issue #10717, reported and patched by leedm777) ........ - -2007-09-14 13:02 +0000 [r82373] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c: Fix DTMF following what has been done in - issue #9401. Thanks irroot. - -2007-09-13 23:12 +0000 [r82359] Jason Parker <jparker@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4 - lines Fix a small typo. retrytime > waittime ........ - -2007-09-13 21:53 +0000 [r82347-82352] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Changed "in" to "queue" in "queue - {pause|unpause} member" command to be more clear. Also added - check to be sure that sixth argument is the word "reason" if full - command is given - - * CHANGES, apps/app_queue.c: Added the ability to pause and unpause - members via the CLI - - * /, apps/app_queue.c: Merged revisions 82346 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep - 2007) | 4 lines Preemptively fixing a possible segfault. It is - possible that queuename is NULL (meaning pause ALL queues), so - use q->name instead. ........ - -2007-09-13 20:13 +0000 [r82345] Jason Parker <jparker@digium.com> - - * /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9 - lines Fix a crash that could occur in cdr_csv when mutliple - threads tried to close the same file. Do we actually need the - locking here? What happens if you open the same file twice, and - two threads try to write to it at the same time? Is fputs() going - to write out the entire line at once? I suspect that it could be - possible for the second fopen to run during the first fputs, so - the position could be in the middle of the previously written - line... Issue 10347, initial patch by explidous (but I removed - all of the paranoia stuff..) ........ - -2007-09-13 19:16 +0000 [r82338-82341] Russell Bryant <russell@digium.com> - - * /, main/astobj2.c: Merged revisions 82339 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) | - 1 line resolve a warning when not building under dev mode - ........ - - * include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged - revisions 82337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) | - 4 lines Only compile in tracking astobj2 statistics if dev-mode - is enabled. Also, when dev mode is enabled, register the CLI - command that can be used to run the astobj2 test and print out - statistics. ........ - -2007-09-13 18:13 +0000 [r82336] Kevin P. Fleming <kpfleming@digium.com> - - * /, LICENSE: Merged revisions 82335 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700 - (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 - Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license - exceptions ........ ................ - -2007-09-13 16:58 +0000 [r82329] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add - setvar support to chan_zap. Just like you can in chan_sip and - chan_iax2 you can now use it with zaptel channels. (done while in - Montreal at the Asterisk bootcamp!) - -2007-09-13 16:27 +0000 [r82327] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 82326 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep - 2007) | 7 lines Added logic to handle the unlikely case that - someone has two queues with the same name. Asterisk will log a - warning message letting the user know that one was already - defined with that name and is it skipping all further instances. - This also will work for realtime queues but in order for that to - happen, the user would have to trigger a perfectly timed reload - as a realtime queue is being looked up, which is highly unlikely - (but taken care of nonetheless). ........ - -2007-09-13 15:26 +0000 [r82321] Russell Bryant <russell@digium.com> - - * include/asterisk/doxyref.h, doc/res_config_sqlite.txt, - res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various - code and documentation cleanups for res_config_sqlite (closes - issue #10711, rbraun_proformatique) - -2007-09-13 15:25 +0000 [r82312-82320] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_jingle.c: Modify rule filters to match with the - Jingle namespace constant - - * include/asterisk/jingle.h: Assign namespace properly - - * channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle - and Jingle DTMF namespaces. As both specifications are in the - Experimental status, the namespaces specified therein shall be of - the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See - the Namespace issuance section in XEP-0053 : - http://www.xmpp.org/extensions/xep-0053.html#namespaces - - * channels/chan_jingle.c: Reflect Jingle DTMF specification changes - -2007-09-13 13:34 +0000 [r82311] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix a missing unref of a member struct. This - was pointed out by Marta. Thanks! This function in 1.4 didn't - have the problem. - -2007-09-13 11:54 +0000 [r82310] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 - Sep 2007) | 4 lines Closes issue #9401, reported and patched by - irrot, with slight modifications by me. Handle DTMF sent by - Asterisk properly. ........ - -2007-09-12 21:57 +0000 [r82297] Russell Bryant <russell@digium.com> - - * /, res/res_agi.c: Merged revisions 82296 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) | - 3 lines Fix a check of the wrong pointer, as pointed out by an - XXX comment left in the code. The problem was harmless, however. - ........ - -2007-09-12 21:55 +0000 [r82294] Jason Parker <jparker@digium.com> - - * channels/chan_iax2.c: After some discussions, we decided that the - return values here were a bit messy. This also fixes a bug on - reload, where peers may not have reregistered properly. - -2007-09-12 21:32 +0000 [r82290-82292] Tilghman Lesher <tlesher@digium.com> - - * /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12 - Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files - ........ - - * main/stdtime/private.h, /, main/stdtime/tzfile.h, - funcs/func_strings.c, apps/app_sms.c, - include/asterisk/localtime.h, main/stdtime/localtime.c: Merged - revisions 82285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007) - | 4 lines Working on issue #10531 exposed a rather nasty 64-bit - issue on ast_mktime, so we updated the localtime.c file from - source. Next we'll have to write ast_strptime to match. ........ - -2007-09-12 21:17 +0000 [r82289] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem - because unless get_member_status returned QUEUE_NORMAL, a NULL - member would be unreferenced. While this didn't cause any crashes - or anything terrible, it still is incorrect - -2007-09-12 20:50 +0000 [r82288] Steve Murphy <murf@digium.com> - - * main/config.c: This fix closes issue #10642 -- it's not perfect, - but should retain most blank lines in config files, via - read/write cycles. - -2007-09-12 20:47 +0000 [r82287] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007) - | 1 line remove a race condition for the creation of - recordthread's, and fix a small memory leak. This closes issue# - 10636 ........ - -2007-09-12 16:24 +0000 [r82283] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build - warnings (closes issue #10698, reported and patched by snuffy) - -2007-09-12 15:53 +0000 [r82279-82282] Russell Bryant <russell@digium.com> - - * utils/hashtest2.c: Change the traversal to use ao2_callback() - instead of an ao2_iterator. Using ao2_callback() is a much more - efficient way of performing an operation on every item in the - container. This change makes hashtest2 run in about 25% of the - time it ran before on my system. In general, I would say that it - makes the most sense to use an ao2_iterator if the operation - being performed is going to take a long time and you don't want - to keep the container locked while you work with each object. - Otherwise, the use of ao2_callback is preferred. - - * /, main/asterisk.c: Merged revisions 82280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) | - 4 lines Clean up the output of "asterisk -h". This tweaks the - wording and wraps lines at 80 characters. (closes issue #10699, - seanbright) ........ - - * /, res/res_agi.c: Merged revisions 82278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) | - 3 lines revert patch from issue #10553, as someone not using - fastagi reported that this broke their system. ........ - -2007-09-12 14:31 +0000 [r82275-82277] Mark Michelson <mmichelson@digium.com> - - * /: Blocking changes from revision 82276 - - * /, apps/app_queue.c: Merged revisions 82274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep - 2007) | 6 lines We should only initialize a realtime queue when - it is allocated, not every time we access it. This prevents the - members ao2_container from being reallocated every time the queue - is accessed. I also removed a debug message I had accidentally - left in on a previous commit. ........ - -2007-09-11 23:07 +0000 [r82273] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Fix to make sure we don't hangup a call when - getting a RLC without sending REL. Found making sure we are Q.784 - (the SS7 test specification) compliant - -2007-09-11 22:38 +0000 [r82269-82270] Russell Bryant <russell@digium.com> - - * main/config.c: remove unused functions that made this file not - build under dev mode - - * /, apps/app_queue.c: Merged revisions 82267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) | - 3 lines Fix incorrect uses of ao2_find(). Every one of these - calls was reading bogus memory ... ........ - -2007-09-11 22:37 +0000 [r82268] Steve Murphy <murf@digium.com> - - * utils/Makefile, main/config.c: This solves an unreported solaris - compile problem (missing -lnsl -lsocket). - -2007-09-11 21:43 +0000 [r82266] Joshua Colp <jcolp@digium.com> - - * /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged - revisions 82265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4 - lines (closes issue #10679) Reported by: andrew Build under dev - mode when K6OPTS is enabled. ........ - -2007-09-11 20:50 +0000 [r82264] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 82263 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) | - 5 lines Fix another missing unref of member objects. This one was - pointed out by Marta. When building the outgoing list in - try_calling(), a member reference is stored in each outgoing - entry. However, when this list got destroyed, the reference was - not released. ........ - -2007-09-11 20:49 +0000 [r82262] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 82261 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1 - line this change should fix issue # 10659 -- what I worry about - is how many other bug reports it may generate. Hopefully, we can - please the/a majority. Hopefully. We shall see. Calls not marked - ANSWERED and with only one channel name will not be posted. This - should eliminate the double CDR's. ........ - -2007-09-11 18:37 +0000 [r82257-82258] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample: Lil' bit more documentation to keep - folks happy. - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes - issue #9433) Reported by: junky Patches: register_trying.diff.txt - uploaded by jcmoore Disable sending 100 Trying on REGISTER - attempts and make it an option. This has been signed off by oej. - -2007-09-11 17:16 +0000 [r82256] Steve Murphy <murf@digium.com> - - * utils/Makefile: fixing up the pthread stuff for hashtest2 - -2007-09-11 16:15 +0000 [r82254] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged - revisions 82249 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) | - 1 line fixed a hold/retrieve issue. ........ - -2007-09-11 16:12 +0000 [r82253] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 82252 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep - 2007) | 6 lines All instances of ao2_iterators which were just - named 'i' have been renamed to 'mem_iter' so that when refcounted - queues are merged into trunk, there will be little confusion - regarding iterator names, especially when a queue and member - iterator are used in the same function. ........ - -2007-09-11 16:05 +0000 [r82251] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) | - 4 lines The sample dundi.conf claims support for a wildcard peer - entry - [*], but the code did not support it. This patch makes it - work. (closes issue #10546, patch by dds, with some changes by - me) ........ - -2007-09-11 15:34 +0000 [r82248] Joshua Colp <jcolp@digium.com> - - * main/cdr.c: (closes issue #10666) Reported by: arkadia Patches: - cdr_lockorder.patch uploaded by arkadia (license 233) Optimize - CDR stuff a bit. - -2007-09-11 15:31 +0000 [r82246-82247] Russell Bryant <russell@digium.com> - - * res/res_agi.c: Remove an unused variable. I have no idea why this - was marked with the unused attribute instead of just removing it. - :) - - * /, res/res_agi.c: Merged revisions 82245 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) | - 9 lines (closes issue #10553) Reported by: juggie Patches: - res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: - juggie When using fastagi, fgets() can return before a full line - is read. Add explicit handling for the case where it gets - interrupted. ........ - -2007-09-11 14:58 +0000 [r82242-82244] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6 - lines (closes issue #10577) Reported by: jamesgolovich Patches: - asterisk-dundifree.diff.txt uploaded by jamesgolovich (license - 176) Don't leak memory when unloading DUNDi. ........ - - * apps/app_meetme.c: (closes issue #10560) Reported by: ruffle - Patches: rb uploaded by ruffle (license 201) Show whether the - conference is locked or not on the CLI. - -2007-09-11 14:35 +0000 [r82237-82241] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 82240 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) | - 2 lines Add a couple more missing unrefs of queue member objects - ........ - - * /, apps/app_queue.c: Merged revisions 82238 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) | - 2 lines Add a missing unref of a queue member in an error - handling block ........ - - * /, apps/app_queue.c: Merged revisions 82236 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) | - 2 lines Document why membercount can not simply be replaced by - ao2_container_count() ........ - -2007-09-11 13:46 +0000 [r82231-82235] Joshua Colp <jcolp@digium.com> - - * utils/Makefile: Include string compatibility file in hashtest2. - - * utils/hashtest2.c: Include compat.h to hopefully make it - compatible with FreeBSD. - - * utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h - exists before including it. - - * main/manager.c: (closes issue #10695) Reported by: junky Patches: - count_showconn.diff uploaded by junky (license 177) Provide a - count of connected users to manager. - - * main/minimime/minimime.c, main/minimime/tests/create.c, - main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes - issue #10692) Reported by: snuffy Patches: minivm.diff uploaded - by snuffy (license 35) Instead of using err (which is not - available under Solaris) use fdprintf with stderr. - -2007-09-10 20:03 +0000 [r82200] Tilghman Lesher <tlesher@digium.com> - - * UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to - have manager-style output, instead of CLI-style output (closes - issue #8254) - -2007-09-10 19:10 +0000 [r82185] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixing a problem where NULL channels would - cause a crash when calling indisposed queue members (i.e. paused, - wrapup time not completed, etc.) - -2007-09-10 18:32 +0000 [r82178] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_queue.c: Merged revisions 82155 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007) - | 2 lines Convert struct member to use refcounts (closes issue - #10199) ........ - -2007-09-10 17:39 +0000 [r82154] Jason Parker <jparker@digium.com> - - * main/db.c: Add a counter to the 'database deltree' CLI command. - Note: this is slightly different than the initial patch, because - I felt that using res <= 0 would be a change in behavior. Closes - issue #10687, patch by junky - -2007-09-10 16:59 +0000 [r82140] Steve Murphy <murf@digium.com> - - * utils/Makefile, utils/hashtest2.c (added): Committing my test for - astobj2, hashtest2.c, along with makefile changes in utils. - -2007-09-10 16:24 +0000 [r82125] Jason Parker <jparker@digium.com> - - * main/db.c: Add counter to 'database show' CLI command. (also a - minor whitespace change that I found along the way) Closes issue - #10683, patch by junky - -2007-09-10 16:19 +0000 [r82124] Steve Murphy <murf@digium.com> - - * main/astobj2.c: Changes applied from marta's team/marta/astobj2 - branch to solve a race condition - -2007-09-10 15:05 +0000 [r82092] Mark Michelson <mmichelson@digium.com> - - * /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 - Sep 2007) | 5 lines Removing non-existent options from misdn - configuration sample. (closes issue #10678, reported and patched - by IgorG) ........ - -2007-09-10 14:26 +0000 [r82062-82077] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: (closes issue #10688) Reported by: casper - Patches: chan_sip.c.82076.diff uploaded by casper (license 55) - Remove double check for zombie flag and optimize things a bit. - - * res/res_agi.c: (closes issue #10684) Reported by: junky Patches: - debug.diff uploaded by junky (license 177) Fix issue with debug - always showing up. - - * apps/app_meetme.c: (closes issue #10686) Reported by: junky - Patches: meet.diff uploaded by junky (license 177) Change NOTICE - message to DEBUG. - -2007-09-09 02:45 +0000 [r82029] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08 - Sep 2007) | 2 lines Fix inline compiles on really old compilers - (who uses gcc 2.7 anymore, really?) (closes issue #10675) - ........ - -2007-09-08 19:01 +0000 [r81998-81999] Russell Bryant <russell@digium.com> - - * include/asterisk/slinfactory.h: Add doxygen documentation for - slinfactory_destroy(), mainly just noting that it doesn't free - the slinfactory itself. (This isn't related to a bug, i'm just - looking over random code) - - * /, main/asterisk.c: Merged revisions 81997 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) | - 2 lines Fix a small memory leak. ast_unregister_atexit() did not - free the entry it removed. ........ - -2007-09-08 16:37 +0000 [r81984] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Make Callerid more consistent in IMAP mail - headers (closes issue #10056, reported and patched by jaroth, - with small modification by me) - -2007-09-08 13:45 +0000 [r81953] Russell Bryant <russell@digium.com> - - * /, .cleancount: Merged revisions 81952 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) | - 11 lines (closes issue #10672) Bump the cleancount so that a - "make clean" will be forced. This is needed because my fix in - revision 81599 made a change to a data structure in file.h, and - since file dependency tracking is only on with dev-mode enabled, - file format modules that don't get rebuilt may crash, as is the - case with this issue. This makes me wonder - how much faster does - the code build without the file dependency tracking enabled? If - it doesn't make much of a difference, then it may be worth just - keeping it on all of the time, or perhaps just not in release - tarballs, so that this type of issue is avoided. ........ - -2007-09-07 19:53 +0000 [r81910-81924] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 81923 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500 - (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable - to be used as the mixmonitor filename. This moves the setting of - the MEMBERINTERFACE variable to before mixmonitor. Issue 10671, - patch by sim. ........ - - * apps/app_queue.c: Add an optional reason parameter to - PauseQueueMember/UnpauseQueueMember applications and manager - events. Issue 8738, patch by rgollent - -2007-09-07 15:29 +0000 [r81891] Mark Michelson <mmichelson@digium.com> - - * /, configs/queues.conf.sample: Merged revisions 81886 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep - 2007) | 3 lines Moving the explanation for joinempty to a more - appropriate place ........ - -2007-09-07 12:32 +0000 [r81858-81873] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Don't check for epoll support when cross - compiling. - - * main/channel.c, main/audiohook.c: Fix memory issue that crept up - with Russell's testing. It is *not* proper to free the frame we - get in ast_write. - -2007-09-06 22:32 +0000 [r81839-81849] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix the build ... oops - - * /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) | - 16 lines (closes issue #9724, closes issue #10374) Reported by: - kenw Patches: 9724.txt uploaded by russell (license 2) Tested by: - kenw, russell Resolve a deadlock that occurs when doing a SIP - transfer to parking. I come across this type of deadlock fairly - often it seems. It is very important to mind the boundary between - the channel driver and the core in respect to the channel lock - and the channel-pvt lock. Channel drivers lock to lock the pvt - and then the channel once it calls into the core, while the core - will do it in the opposite order. The way this is avoided is by - having channel drivers either release their pvt lock while - calling into the core, or such as in this case, unlocking the pvt - just long enough to acquire the channel lock. ........ - -2007-09-06 22:06 +0000 [r81827] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 81826 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1 - line We added COPTS for ASTCFLAGS additions, but not LDOPTS for - ASTLDFLAGS. This adds LDOPTS ........ - -2007-09-06 21:01 +0000 [r81814] Joshua Colp <jcolp@digium.com> - - * channels/iax2-parser.c: Initialize iax_frames variable to NULL, - keeps valgrind happy. - -2007-09-06 20:54 +0000 [r81783-81813] Russell Bryant <russell@digium.com> - - * CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE() - function that can retrieve the state of an extension that has a - hint. (closes issue #10635, adamgundy) - - * CHANGES: s/DEVSTATE/DEVICE_STATE/ - - * funcs/func_devstate.c: Rename the DEVSTATE() function to - DEVICE_STATE() to better conform to how other functions are - named. (inspired by issue #10635) - - * CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function - from my sandbox branch into trunk. This function will let you - retrieve the list of devices or name associated with a hint. - (inspired by issue #10635) - -2007-09-06 20:16 +0000 [r81782] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c, CHANGES: (closes issue #10377) Reported - by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak - (license 7) Add skinny show device, skinny show line, and skinny - show settings CLI commands. - -2007-09-06 20:05 +0000 [r81781] Russell Bryant <russell@digium.com> - - * configs/extensions.conf.sample: Fix the syntax of declaring a - hint with a name to be compatible with trunk - -2007-09-06 20:00 +0000 [r81779] Jason Parker <jparker@digium.com> - - * /, include/asterisk/astobj2.h: Merged revisions 81778 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2 - lines This should fix a build issue that people building against - uClibc were seeing with the addition of astobj2 ........ - -2007-09-06 19:43 +0000 [r81777] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 - lines (closes issue #10122) Reported by: stevefeinstein Patches: - meetme-unmute-manager.diff uploaded by qwell (license 4) Tested - by: stevefeinstein After looking over the code I agree with - Qwell. Setting the file descriptor to conference each time just - causes a fight back and forth. ........ - -2007-09-06 17:00 +0000 [r81745] Philippe Sultan <philippe.sultan@gmail.com> - - * /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged - revisions 81743 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) - | 1 line Various string length fixes. Removed an unused variable - in aji_client structure (context) ........ - -2007-09-06 16:57 +0000 [r81744] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/safe_asterisk: Incorporate the ability to log - output of safe_asterisk to syslog (closes issue #9882) - -2007-09-06 16:38 +0000 [r81742] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Patch on 10575. Add support for unequipped - CIC (UCIC) message as well as improve some of our CIC flags in - chan_zap - -2007-09-06 16:31 +0000 [r81730] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81713 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep - 2007) | 6 lines Fixes an issue where valid DTMF had to be pressed - twice to exit a queue if a member's phone was ringing. (closes - issue #10655, reported by strider2k, patched by me) ........ - -2007-09-06 15:43 +0000 [r81712] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/astobj2.h, main/astobj2.c: various changes to - the documentation, and redefinition of ao2_hash_fn and - ao2_callback_fn typedefs, in preparation to more cleanup of the - _search_flags Please do not merge this change to 1.4 yet - there - are no functional changes anyways. - -2007-09-06 15:21 +0000 [r81683] Mark Michelson <mmichelson@digium.com> - - * /, res/res_features.c: Merged revisions 81682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep - 2007) | 5 lines Fixes a memory leak (closes issue #10658, - reported and patched by Ivan) ........ - -2007-09-06 14:24 +0000 [r81651] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 81650 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007) - | 3 lines According to both RFC 3920 - section 9.1.2 - and - Google's XMPP server complaint, if set, the 'from' attribute must - be set to the user's full JID. ........ - -2007-09-05 21:59 +0000 [r81632] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Not having this epoll specific code in - wait_for_answer was causing app_queue to infinitely loop. This - makes it so it doesn't. Thanks to file for pointing out where the - problem was and showing a similar function in app_dial as an - example of how to fix it. - -2007-09-05 21:45 +0000 [r81631] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05 - Sep 2007) | 2 lines Solaris x86 compatibility fix ........ - -2007-09-05 20:58 +0000 [r81601] Dwayne M. Hubbard <dhubbard@digium.com> - - * apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller - -2007-09-05 20:58 +0000 [r81600] Russell Bryant <russell@digium.com> - - * include/asterisk/file.h, /, main/say.c, res/res_features.c, - main/file.c, include/asterisk/channel.h: Merged revisions 81599 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | - 11 lines Fix an issue that can occur when you do an attended - transfer to parking. If you complete the transfer before the - announcement of the parking spot finishes, then the channel being - parked will hear the remainder of the announcement. These changes - make it so that will not happen anymore. Basically, res_features - sets a flag on the channel is playing the announcement to so that - the file streaming core knows that it needs to watch out for a - channel masquerade, and if it occurs, to abort the announcement. - (closes BE-182) ........ - -2007-09-05 16:48 +0000 [r81568] Tilghman Lesher <tlesher@digium.com> - - * utils: Add two more generated files (requested by mvanbaak via - irc) - -2007-09-05 16:31 +0000 [r81560] Jason Parker <jparker@digium.com> - - * include/asterisk/devicestate.h, res/res_config_odbc.c, - channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c, - res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c, - include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c, - res/res_config_sqlite.c, include/asterisk/sha1.h, - include/asterisk/stringfields.h, include/asterisk/features.h: - Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy - -2007-09-05 15:32 +0000 [r81526-81535] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Weird. When I merged my changes from 1.4, they - merged into the wrong function. This should fix the build for - trunk. - - * /, apps/app_queue.c: Merged revisions 81525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep - 2007) | 4 lines Fixing the build... ........ - -2007-09-05 15:16 +0000 [r81524] Jason Parker <jparker@digium.com> - - * channels/chan_phone.c, /: Merged revisions 81523 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500 - (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL - channel tech. Also changed load_module function to use defines - rather than numbers for return values. Issue 10651, patch by - rbraun_proformatique, with additions by me. ........ - -2007-09-05 15:04 +0000 [r81522] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81520 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep - 2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only - count members who are logged in and available. (related to issue - #10652, reported by wuwu) ........ - -2007-09-05 14:47 +0000 [r81519] Steve Murphy <murf@digium.com> - - * include/asterisk/config.h, main/config.c: this set of changes - fixes issue # 10643 by keeping track of the last object defined - in a file, and attaching any accumulated comments to that object - (category header or variable declaration). The file_save routine - also had to be upgraded to output these trailing comments. - Config.h was modified to include the trailing comment list on - categories and variables. - -2007-09-05 13:13 +0000 [r81459-81493] Joshua Colp <jcolp@digium.com> - - * main/editline/sys.h: Finish up commit from revision 81452 by - removing last remnants of strlcat/strlcpy checks. - -2007-09-04 20:59 +0000 [r81454-81456] Jason Parker <jparker@digium.com> - - * /, apps/app_followme.c: Merged revisions 81455 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500 - (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file, - we can just check whether it exists. Issue 10634, patch by me, - testing by pabelanger, sanity checked by bweschke ........ - - * /, configs/followme.conf.sample: Merged revisions 81453 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500 - (Tue, 04 Sep 2007) | 4 lines Change default followme config file - to point to the correct files. Issue 10644, patch by pabelanger - ........ - -2007-09-04 19:51 +0000 [r81445-81452] Russell Bryant <russell@digium.com> - - * main/editline/configure, main/editline/configure.in: Don't check - for and include strlcpy and strlcat in editline. We also include - them directly in Asterisk. For platforms that need them (like my - mac), you will get a linker error due to the functions being - included twice. - - * /, include/asterisk/astobj2.h, channels/chan_iax2.c, - main/astobj2.c: Merged revisions 81448 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | - 4 lines Remove the typedefs on ao2_container and ao2_iterator. - This is simply because we don't typedef objects anywhere else in - Asterisk, so we might as well make this follow the same - convention. ........ - - * include/asterisk/logger.h: logger.h depends on options.h, so go - ahead and include it - -2007-09-04 16:41 +0000 [r81443] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007) - | 2 lines there is no point in sending 401 Unauthorized to a UAS - that sent us a properly-formatted Authentication header with the - expected username and nonce but an incorrect response (which - indicates the shared secret does not match)... instead, let's - send 403 Forbidden so that the UAS doesn't retry with the same - authentication credentials repeatedly ........ - -2007-09-04 14:28 +0000 [r81436-81441] Joshua Colp <jcolp@digium.com> - - * configs/extensions.ael.sample: (closes issue #10633) Reported by: - pabelanger Patches: extensions.ael.sample.patch uploaded by - pabelanger (license 224) Update extensions.ael.sample with - voicemail and | changes. - - * /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6 - lines (closes issue #10632) Reported by: jamesgolovich Patches: - asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich - (license 176) Fix memory leak when unloading chan_iax2. The - firmware files were not being freed. ........ - - * main/channel.c, /: Merged revisions 81437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4 - lines (closes issue #10476) Reported by: mdu113 Only look for the - end of a digit when waiting for a digit. This in turn disables - emulation in the core. ........ - - * /, main/dns.c: Merged revisions 81435 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7 - lines (closes issue #10610) Reported by: john Patches: - dns.c.patch uploaded by john (license 218) Tested by: mvanbaak - Don't return a match if no SRV record actually exists. ........ - -2007-09-03 18:59 +0000 [r81434] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) | - 5 lines Remove a couple of calls to ast_string_field_free_pools() - on peers in error handling blocks in the code for building peers. - The peer object destructor does this and doing it twice will - cause a crash. (closes issue #10625, reported by and patched by - pnlarsson) ........ - -2007-09-03 18:01 +0000 [r81430-81432] Tilghman Lesher <tlesher@digium.com> - - * main/config.c: Once we get past the file checks, we're loading, - so clear the FILEUNCHANGED flag (fixes #include) (closes issue - #10629) - - * /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007) - | 2 lines The IF() function was not allowing true values that had - embedded colons (closes issue #10613) ........ - - * main/config.c: We shouldn't use a filename blindly without - checking to make sure it's unused first - -2007-09-01 06:03 +0000 [r81427] Mark Michelson <mmichelson@digium.com> - - * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions - 81426 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep - 2007) | 4 lines Making match_by_addr into ao2_match_by_addr and - making it available everywhere since it could be a handy callback - to have ........ - -2007-08-31 21:29 +0000 [r81419] Russell Bryant <russell@digium.com> - - * /, include/asterisk/astobj2.h: Merged revisions 81418 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) | - 2 lines Remove references to a debugging parameter that does not - exist ........ - -2007-08-31 19:50 +0000 [r81417] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81416 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug - 2007) | 6 lines Fixed broken behavior of a reload on realtime - queues. Prior to this patch, if a reload was issued and a - realtime queue had callers waiting in it, then the queue would be - removed from the queue list, but it would not actually be freed - (in fact, a debug message warning about a memory leak would come - up). With this patch, reloads do not touch realtime queues at - all. ........ - -2007-08-31 18:46 +0000 [r81413] Jason Parker <jparker@digium.com> - - * apps/app_dial.c, /: Merged revisions 81412 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 - (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line - with the existing alpha order. Issue 10621, initial patch by - junky ........ - -2007-08-31 17:43 +0000 [r81411] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 - Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command - available. Closes issue 10548, reported by keepitcool. ........ - -2007-08-31 15:58 +0000 [r81408] Kevin P. Fleming <kpfleming@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007) - | 2 lines add missing "transcoder show" (and deprecated "show - transcoder") CLI commands that were in 1.2 but never added to 1.4 - ........ - -2007-08-31 15:54 +0000 [r81402-81407] Joshua Colp <jcolp@digium.com> - - * /, res/res_speech.c: Merged revisions 81406 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2 - lines Make it the engine's responsible to check for the presence - of results. ........ - - * /, res/res_features.c: Merged revisions 81403 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4 - lines (closes issue #10618) Reported by: dimas Don't pass through - the stopped sounds frame.... just drop it. ........ - - * /, res/res_features.c: Merged revisions 81401 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4 - lines (closes issue #10009) Reported by: dimas Don't output a - bridge failed warning message if it failed because one of the - channels was part of the masquerade process. That is perfectly - normal. ........ - -2007-08-30 23:52 +0000 [r81400] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c: Add new queryable fields from zaptel to 'zap - show status' - -2007-08-30 22:08 +0000 [r81398] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug - 2007) | 7 lines Removing an extraneous (and possibly misleading) - log message. Firstly, if the announce file isn't found, the - streaming functions will report it. Secondly, not all non-zero - returns from play_file mean that the announce file wasn't found. - Positive return values simply mean that a digit was pressed (most - likely to skip through the announcement). (closes issue #10612, - reported and patched by dimas) ........ - -2007-08-30 21:25 +0000 [r81394-81396] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6 - lines (closes issue #10514) Reported by: casper Patches: - chan_sip.c.80129.diff uploaded by casper (license 55) Remove - needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it - to ever be that value. ........ - - * channels/chan_sip.c: (closes issue #10565) Reported by: tootai - Make sure the external IP address has the standard SIP port set - for when the user does not specify the port in the externip - setting. - -2007-08-30 21:16 +0000 [r81393] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 81392 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1 - line via issue 10599, where 'CDR already initialized' messages - are being generated. Since all channels will have an init'd CDR - attached at creation time, this message is now particularly - useless. Removed. ........ - -2007-08-30 20:55 +0000 [r81391] Joshua Colp <jcolp@digium.com> - - * apps/app_minivm.c: (closes issue #10336) Reported by: junky - Patches: minivm_output2.diff uploaded by junky (license 177) - Change console output of minivm show stats to be more simple for - external parsing. - -2007-08-30 20:31 +0000 [r81389-81390] Tilghman Lesher <tlesher@digium.com> - - * main/sched.c: A schedule id of 0 is not possible and is used to - flag that we want to add a new item - - * apps/app_readexten.c: Change wording as requested by Kevin - -2007-08-30 18:52 +0000 [r81388] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample: Added note to sample queues.conf file - to line up with most recent change regarding setinterfacevar. - MEMBERREALTIME indicates whether a member is realtime. - -2007-08-30 17:51 +0000 [r81387] Tilghman Lesher <tlesher@digium.com> - - * main/logger.c: Always force reread of the config when we're - rotating the log file (closes issue #10598) - -2007-08-30 15:40 +0000 [r81384] Russell Bryant <russell@digium.com> - - * /, channels/h323/ast_h323.cxx: Merged revisions 81383 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) | - 3 lines Add missing checks for the PTRACING define. (closes issue - #10559, paravoid) ........ - -2007-08-30 15:36 +0000 [r81382] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug - 2007) | 3 lines Changed some manager event messages to reflect - whether a queue member is a realtime member or not ........ - -2007-08-30 15:34 +0000 [r81380] Russell Bryant <russell@digium.com> - - * configs/modem.conf.sample (removed), /, configs/enum.conf.sample, - configs/extensions.ael.sample: Merged revisions 81379 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) | - 3 lines Fix a typo, update a reload command, and remove an unused - configuration file. (closes issue #10606, casper) ........ - -2007-08-30 15:24 +0000 [r81378] Tilghman Lesher <tlesher@digium.com> - - * apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN - function (closes issue #10082) - -2007-08-30 14:54 +0000 [r81376] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30 - Aug 2007) | 1 line Fixed some warnings. ........ - -2007-08-30 14:42 +0000 [r81370-81372] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls - Patches: pbx.diff uploaded by jmls (license 141) Add REASON - dialplan variable for when an originated call fails and the - failed extension is executed. - - * /, res/res_features.c: Merged revisions 81369 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4 - lines (issue #10599) Reported by: dimas Handle the -1 control - subclass during feature dialing (it indicates to stop sounds). - ........ - -2007-08-30 08:50 +0000 [r81368] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 81367 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) | - 11 lines Fixed a severe issue where a misdn_read would lock the - channel, but read would not return because it blocks. later - chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER - which could result in a deadlock situation. misdn_read will now - not block forever anymore, and we don't queue the ANSWER frame at - all when we already was called with misdn_answer -> answer would - be called twice. Also we don't explicitly send a RELEASE_COMPLETE - on receiption of a RELEASE anymore, because mISDN does that for - us, this resulted in a problem on some switches, which would - block our port after some calls for a short while. ........ - -2007-08-29 22:05 +0000 [r81365] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Added the MEMBERREALTIME variable when using - setinterfacevar in queues.conf - -2007-08-29 21:55 +0000 [r81364] Joshua Colp <jcolp@digium.com> - - * include/asterisk/event.h: Make the event header file work under - C++. - -2007-08-29 21:30 +0000 [r81363] Steve Murphy <murf@digium.com> - - * main/config.c: init newer so compile won't complain. - -2007-08-29 21:25 +0000 [r81362] Russell Bryant <russell@digium.com> - - * main/config.c: make trunk build again. murf will have to review - this to see if it was the right fix, as it is related to his last - change. - -2007-08-29 20:55 +0000 [r81361] Steve Murphy <murf@digium.com> - - * res/res_config_pgsql.c, channels/chan_sip.c, - include/asterisk/config.h, channels/chan_iax2.c, - channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c, - main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c, - main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, - main/http.c, utils/extconf.c, apps/app_directory.c, - apps/app_parkandannounce.c, apps/app_voicemail.c: This code was - in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I - didn't add it to 1.4 yet, because it's not entirely clear to me - if this is a bug fix or an enhancement. A lot of files were - affected by small changes like ast_variable_new getting an added - arg, for the file name the var was defined in; ast_category_new - gets added args of filename and lineno; ast_category and - ast_variable structures now record file and lineno for each - entry; a list of all #include and #execs in a config file (or any - of its inclusions are now kept in the ast_config struct; at save - time, each entry is put back into its proper file of origin, in - order. #include and #exec directives are folded in properly. - Headers indicating that the file was generated, are generated - also for each included file. Some changes to main/manager.c to - take care of file renaming, via the UpdateConfig command. - Multiple inclusions of the same file are handled by exploding - these into multiple include files, uniquely named. There's - probably more, but I can't remember it right now. - -2007-08-29 19:41 +0000 [r81353-81356] Russell Bryant <russell@digium.com> - - * main/event.c: Try to clarify the rules on changing ast_event and - ast_event_ie - - * main/event.c: Fix parenthesis from my last commit - - * main/event.c: Change pointer aritmetic on void * to char * - - * main/event.c: there is not actually code that sends these over - the network in trunk yet - -2007-08-29 16:39 +0000 [r81350] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug - 2007) | 12 lines This patch, in essence, will correctly pause a - realtime queue member and reflect those changes in the realtime - engine. (issue #10424, reported by irroot, patch by me) This - patch creates a new function called update_realtime_member_field, - which is a generic function which will allow any one field of a - realtime queue member to be updated. This patch only uses this - function to update the paused status of a queue member, but it - lays the foundation for persisting the state of a realtime member - the same way that static members' state is maintained when using - the persistentmembers setting ........ - -2007-08-29 16:25 +0000 [r81348] Joshua Colp <jcolp@digium.com> - - * main/event.c: Return ast_event_get_ie_raw to using an iterator - and fix logic in ast_event_iterator_next. - -2007-08-29 16:09 +0000 [r81347] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81346 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug - 2007) | 3 lines Changed some tabs to spaces ........ - -2007-08-29 16:07 +0000 [r81344-81345] Joshua Colp <jcolp@digium.com> - - * main/event.c: This concludes bringing trunk back to a working - state. - - * include/asterisk/event.h, main/event.c: To keep others happy... - revert part of my additions so trunk works. - -2007-08-29 15:59 +0000 [r81343] Russell Bryant <russell@digium.com> - - * /, main/Makefile: Merged revisions 81342 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | - 3 lines If chan_h323 is not being built, don't use g++ to do the - final link of Asterisk. (in response to a question on the - asterisk-dev list) ........ - -2007-08-29 15:57 +0000 [r81341] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81340 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug - 2007) | 8 lines This fix creates a more accurate way of detecting - whether realtime members were deleted. (closes issue 10541, - reported by Alric, patched by me) The REALLY nice things about - this patch is that queue members now have a "realtime" field - which will be true if the member is a realtime member. This means - we can check this value prior to certain processing if it should - ONLY be done for realtime members. ........ - -2007-08-29 15:21 +0000 [r81335] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Changed one too many variable settings in - issue #9315 (closes issue #10592) - -2007-08-29 15:19 +0000 [r81334] Joshua Colp <jcolp@digium.com> - - * include/asterisk/event.h, include/asterisk/event_defs.h, - main/event.c: Add API calls for iterating through an event. This - should allow events to have multiple information elements (while - there was nothing preventing it before you could not actually - access any except the first one). - -2007-08-29 14:19 +0000 [r81333] Mark Michelson <mmichelson@digium.com> - - * apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue - #10591, reported and patched by junky, with small modification by - me) - -2007-08-29 14:16 +0000 [r81326-81332] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 - lines (closes issue #9690) Reported by: mattv Make rtp timeouts - work even if two RTP streams are directly bridged in the RTP - stack. ........ - - * include/asterisk/utils.h: Add inline function for signed linear - subtraction. - -2007-08-28 21:39 +0000 [r81292] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) | - 3 lines Change the message about receiving a mini-frame before - the first full voice frame to a DEBUG message. ........ - -2007-08-28 21:35 +0000 [r81290] Joshua Colp <jcolp@digium.com> - - * main/logger.c: Add some read/write locking magic to make logger - reload operate again. - -2007-08-28 20:03 +0000 [r81277] Tilghman Lesher <tlesher@digium.com> - - * main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support - better rotation of log files to be more like system logging - (closes issue #10398) - -2007-08-28 19:12 +0000 [r81227-81264] Russell Bryant <russell@digium.com> - - * include/asterisk/audiohook.h: Change the audiohook lock and - unlock wrappers to macros instead of inline functions. As inline - functions, the lock debug information will show that these are - always locked in audiohooks.h instead of the file where the lock - was actually acquired. - - * funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking - around the uses of datastore_add and _find. There are still more - places in the tree that I have not yet changed if someone wants - to go through and find the places they are used without the - channel locked. - - * main/channel.c, funcs/func_volume.c, include/asterisk/channel.h: - * Constify the uid field of channel datastores * Convert some - spaces to tabs in func_volume * Add a note in channel.h making it - clear that none of the datastore API calls lock the channel they - are given, so the channel should be locked before calling the - functions that take a channel argument. - - * include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c, - doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by: - nic_bellamy Patches: - 2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by - nic_bellamy (license 213) Add support for configurable file - locking methods. The default is "lockfile", which is the old - behavior. There is an additional option, "flock", which is - intended for use in situations where the lockfile method will not - work, such as with SMB/CIFS mounts. - - * /, configs/indications.conf.sample: Merged revisions 81226 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) | - 2 lines Add Russian tones. (closes issue #7953, hanabana) - ........ - -2007-08-28 14:37 +0000 [r81210] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10579) Reported by: ornati - Make sure the called channel during the attended transfer process - becomes associated with the calling channel so that the - ast_waitfor_* call works properly under epoll. - -2007-08-28 14:12 +0000 [r81121-81190] Mark Michelson <mmichelson@digium.com> - - * /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28 - Aug 2007) | 5 lines Fixes a forwarding problem when using - res_config_mysql (closes issue #10573, reported by chrisvaughan, - patch suggested by chrisvaughan as well) ........ - - * /, apps/app_queue.c: Merged revisions 81158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug - 2007) | 5 lines Resolve a potential deadlock. In this case, a - single queue is locked, then the queue list. In changethread(), - the queue list is locked, and then each individual queue is - locked. Under the right circumstances, this could deadlock. As - such, I have unlocked the individual queue before locking the - queue list, and then locked the queue back after the queue list - is unlocked. ........ - - * /, channels/chan_agent.c: Merged revisions 81120 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27 - Aug 2007) | 7 lines DTMF begin frames should be ignored so that - when an agent acks a call with the '#' key, he doesn't cause a - queue's announce file to be interrupted. Also went ahead and did - the same for the '*' key and for ending a call. (closes issue - #10528, reported by deskhack, patched by me) ........ - -2007-08-27 20:55 +0000 [r81118] Tilghman Lesher <tlesher@digium.com> - - * apps/app_directed_pickup.c: Enhance Pickup to do native - pickupgroup pickup when no arguments are specified (closes issue - #10404) - -2007-08-27 17:44 +0000 [r81043-81098] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh - well ... it's harmless. Merged revisions 81065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) | - 1 line explicity define a variable as a boolean ........ - - * /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) | - 3 lines Add a \todo to note that this module leaks most of the - memory it allocates on unload and should be fixed (when I'm not - in the middle of something else ...). ........ - - * /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27 - Aug 2007) | 11 lines (closes issue #10419) Reported by: - mustardman Patches: asterisk-mohposition.diff.txt uploaded by - jamesgolovich (license 176) This patch fixes a few problems with - music on hold. * Fix issues with starting at the beginning of a - file when it shouldn't. * Fix the inuse counter to be decremented - even if the class had not been set to be deleted when not in use - anymore * Don't arbitrarily limit the number of MOH files to 255 - ........ - -2007-08-27 15:03 +0000 [r81013] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6 - lines (closes issue #10561) Reported by: jesselang Patches: - chan_sip-ChannelReload-20080825.patch uploaded by jesselang - (license 202) Remove an extra \r\n to make the ChannelReload - event conform with every other event. ........ - -2007-08-27 14:56 +0000 [r81011] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 81010 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug - 2007) | 3 lines Found a case where the queue's membercount is - off. It does not take into account dynamic members on a reload. - ........ - -2007-08-27 13:35 +0000 [r80962-80991] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Remove places that say if no language is - specified it will default to english... since on some setups this - is untrue. - - * /, main/rtp.c: Merged revisions 80974 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4 - lines (closes issue #10562) Reported by: idkpmiller Correct - jitter value output in the CLI to be as expected. ........ - - * configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG - Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20) - Fix up sip.conf sample configuration. - -2007-08-26 18:12 +0000 [r80933] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) | - 3 lines Remove an extra signal_condition() for the scheduler - thread. (closes issue #10564, patch from casper) ........ - -2007-08-25 17:55 +0000 [r80821-80898] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) | - 7 lines Fix some issues with the handling of the scheduler in - chan_iax2. Most of the places that scheduled items to be executed - by the scheduler thread did not signal the scheduler thread to - wake up so that it could recalculate the time until the next - action. These changes will make the scheduler thread more - responsive and ensure that actions get executed as close to when - intended as possible instead of it being possible for very long - delays. ........ - - * pbx/pbx_dundi.c: localize a variable and remove a duplicate error - message - - * apps/app_queue.c: use ast_strlen_zero - - * /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) | - 5 lines If dnsmgr is in use, and no DNS servers are available - when Asterisk first starts, then don't give up on poking peers. - Allow the poke to get rescheduled so that it will work once the - dnsmgr is able to resolve the host. (closes issue #10521, patch - by jamesgolovich) ........ - - * /, main/dsp.c: Merged revisions 80820 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) | - 7 lines Improve the debouncing logic in the DTMF detector to fix - some reliability issues. Previously, this code used a shift - register of hits and non-hits. However, if the start of the digit - isn't clean, it is possible for the leading edge detector to miss - the digit. These changes replace the flawed shift register logic - and also does the debouncing on the trailing edge as well. - (closes issue #10535, many thanks to softins for the patch) - ........ - -2007-08-24 20:21 +0000 [r80819] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Merged revisions 80818 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007) - | 3 lines A minor correction to the available logic of autofill. - If a queue member is paused, they're not really "available" so - don't count them as such. Somewhat related to issue #10155 - ........ - -2007-08-24 19:50 +0000 [r80817] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Fix documentation for Set (closes issue #10549) - -2007-08-24 19:03 +0000 [r80790] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 80789 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1 - line From a complaint by jmls, I realize that the message in - cdr_disposition is unnecessary. To get failure disposition, just - return -1; no use having more than one case do that. ........ - -2007-08-24 18:05 +0000 [r80778] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add VMWI chan_zap support #9909 - -2007-08-24 15:53 +0000 [r80751] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug - 2007) | 3 lines Fix a possible crash in IMAP voicemail. ........ - -2007-08-24 15:42 +0000 [r80748] Steve Murphy <murf@digium.com> - - * utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well - -2007-08-24 15:29 +0000 [r80725] Russell Bryant <russell@digium.com> - - * /, utils/ael_main.c: Merged revisions 80722 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) | - 3 lines Tweak the formatting of this MODULEINFO block. I think - this would have caused a "*" to get in the menuselect-tree file. - ........ - -2007-08-24 14:55 +0000 [r80690-80718] Steve Murphy <murf@digium.com> - - * /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80717 | murf | 2007-08-24 08:48:49 -0600 (Fri, 24 Aug 2007) | 1 - line This change addresses JerJer's complaint that aelparse - builds and installs even if pbx_ael is unchecked in the - menuselect stuff. ........ - -2007-08-24 11:49 +0000 [r80662] Philippe Sultan <philippe.sultan@gmail.com> - - * /, channels/chan_gtalk.c: Merged revisions 80661 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 - Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are - answered too early, which results in CDRs wrongly stating that a - call was ANSWERED when the calling party cancelled a call before - before being established. We must not answer the call upon - reception of a 'transport-accept' iq packet, but this packet - still needs to be acknowledged, otherwise the remote peer would - close the call (like in #8970). ........ - -2007-08-23 23:37 +0000 [r80649] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, - res/ael/ael.y, res/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-test7: an unreported crash I debugged, - looked like it was backing up way too far after hitting the - syntax error. An inspection of the code revealed that error - tokens in lists were not rearranged when the rules were - rearranged as part of a code neatening-up process. By moving the - error tokens to where they should be, I also reduced the number - of shift/reduce conflicts to 3 instead of 8. This introduces - subtle differences in error messages, so the regressions had to - be updated. - -2007-08-23 21:34 +0000 [r80510-80616] Russell Bryant <russell@digium.com> - - * apps/app_while.c: Use the comma separator in app_while. reported - by blitzrage on irc, patched by me - - * /, res/res_features.c, include/asterisk/features.h: Merged - revisions 80573 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | - 5 lines When executing a dynamic feature, don't look it up a - second time by digit pattern after we already looked it up by - name. This causes broken behavior if there is more than one - feature defined with the same digit pattern. (closes issue - #10539, reported by bungalow, patch by me) ........ - - * /, funcs/func_timeout.c: Merged revisions 80547 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80547 | russell | 2007-08-23 14:29:44 -0500 (Thu, 23 Aug 2007) | - 3 lines Revert very broken fix for issue #10540 ... none of these - values take ms so I don't know what I was thinking ........ - - * /, funcs/func_timeout.c: Merged revisions 80539 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80539 | russell | 2007-08-23 14:21:53 -0500 (Thu, 23 Aug 2007) | - 4 lines Fix func_timeout to take values in floating point so 1.5 - actually means 1.5 seconds instead of being rounded. (closes - issue #10540, reported by spendergrass, patch by me) ........ - - * doc/asterisk-mib.txt, res/snmp/agent.c: Fix a typo in the - Asterisk MIB and fix astNumChanBridged so it acts as a counter - again (closes issue #10118, patch by jeffg) - -2007-08-23 17:18 +0000 [r80508] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 80501 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80501 | kpfleming | 2007-08-23 12:08:25 -0500 (Thu, 23 Aug 2007) - | 2 lines report the actual channel number that was unregistered, - instead of assuming that the interface list consists of channels - 1 through <x> with no gaps in the sequence ........ - -2007-08-23 17:04 +0000 [r80470-80500] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 80499 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80499 | russell | 2007-08-23 12:02:50 -0500 (Thu, 23 Aug 2007) | - 3 lines Fix some code where it was possible for a reference to a - peer to not get released when it should. Thank you to Marta - Carbone for pointing this out! ........ - - * /, res/res_agi.c: Merged revisions 80469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80469 | russell | 2007-08-23 10:49:28 -0500 (Thu, 23 Aug 2007) | - 2 lines Revert res_agi fix that didn't quite work until we get it - right ... ........ - -2007-08-23 15:48 +0000 [r80453-80468] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: If no default language has been specified - print out that it will default to english when using sip show - peer or sip show user. - - * main/minimime/mm.h: Return trunk to a working state by including - compat.h in minimime. - -2007-08-22 23:26 +0000 [r80428-80429] Jason Parker <jparker@digium.com> - - * main/minimime/mm_util.c, main/minimime/mm_codecs.c, - main/minimime/mm_mem.h, main/minimime/mm_base64.c, - main/minimime/mm.h: Convert minimime to use the proper uint*_t - types, rather than u_int*_t - - * apps/app_minivm.c: Cast calls to getpid. This was done in 1.4 - already, this one was just new - -2007-08-22 22:54 +0000 [r80361-80427] Russell Bryant <russell@digium.com> - - * /, include/asterisk/astobj2.h: Merged revisions 80426 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) | - 6 lines Add some more documentation on iterating ao2 containers. - The documentation implies that is possible to miss an object or - see an object twice while iterating. After looking through the - code and talking with mmichelson, I have documented the exact - conditions under which this can happen (which are rare and - harmless in most cases). ........ - - * /, main/astobj2.c: Merged revisions 80424 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) | - 10 lines When converting this code to use the list macros, I - changed it so objects are added to the head of a bucket instead - of the tail. However, while looking over code with mmichelson, we - noticed that the algorithm used in ao2_iterator_next requires - that items are added to the tail. This wouldn't have caused any - huge problem, but it wasn't correct. It meant that if an object - was added to a container while you were iterating it, and it was - added to the same bucket that the current element is in, then the - new object would be returned by ao2_iterator_next, and any other - objects in the bucket would be bypassed in the traversal. - ........ - - * channels/chan_iax2.c: allow peers and users to go into a hash - table - - * /, channels/chan_sip.c: Merged revisions 80390 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80390 | russell | 2007-08-22 16:00:44 -0500 (Wed, 22 Aug 2007) | - 3 lines Don't crash when using realtime in chan_sip without an - insecure setting in the database. (closes issue #10348, reported - by link55, fixed by me) ........ - - * channels/chan_iax2.c: Unsubscribe from MWI events in the peer - destructor - - * /, main/Makefile, include/asterisk/astobj2.h (added), - include/asterisk/strings.h, channels/chan_iax2.c, main/astobj2.c - (added): Merged revisions 80362 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | - 34 lines Merge changes from team/russell/iax_refcount. This set - of changes fixes problems with the handling of iax2_user and - iax2_peer objects. It was very possible for a thread to still - hold a reference to one of these objects while a reload operation - tries to delete them. The fix here is to ensure that all - references to these objects are tracked so that they can't go - away while still in use. To accomplish this, I used the astobj2 - reference counted object model. This code has been in one of - Luigi Rizzo's branches for a long time and was primarily - developed by one of his students, Marta Carbone. I wanted to go - ahead and bring this in to 1.4 because there are other problems - similar to the ones fixed by these changes, so we might as well - go ahead and use the new astobj if we're going to go through all - of the work necessary to fix the problems. As a nice side benefit - of these changes, peer and user handling got more efficient. - Using astobj2 lets us not hold the container lock for peers or - users nearly as long while iterating. Also, by changing a define - at the top of chan_iax2.c, the objects will be distributed in a - hash table, drastically increasing lookup speed in these - containers, which will have a very big impact on systems that - have a large number of users or peers. The use of the hash table - will be made the default in trunk. It is not the default in 1.4 - because it changes the behavior slightly. Previously, since peers - and users were stored in memory in the same order they were - specified in the configuration file, you could influence peer and - user matching order based on the order they are specified in the - configuration. The hash table does not guarantee any order in the - container, so this behavior will be going away. It just means - that you have to be a little more careful ensuring that peers and - users are matched explicitly and not forcing chan_iax2 to have to - guess which user is the right one based on secret, host, and - access list settings, instead of simply using the username. If - you have any questions, feel free to ask on the asterisk-dev - list. ........ - - * /, res/res_agi.c: Merged revisions 80360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | - 5 lines Juggie in #asterisk-dev was reporting problems where - fgets would return without reading the whole line when using - fastagi. When this happens, errno was set to EINTR or EAGAIN. - This patch accounts for the possibility and lets fgets continue - in that case. ........ - -2007-08-22 18:54 +0000 [r80303-80331] Jason Parker <jparker@digium.com> - - * Makefile, build_tools/mkpkgconfig, /, build_tools/make_build_h, - build_tools/strip_nonapi, build_tools/prep_moduledeps, - build_tools/make_buildopts_h: Merged revisions 80330 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r80330 | qwell | 2007-08-22 13:53:18 -0500 (Wed, 22 Aug - 2007) | 7 lines Fix a few build issues in Solaris (and likely - others). Use GREP and ID variables from autoconf. Reported to me - in #asterisk-dev I forgot who reported this - sorry. :( ........ - - * Makefile, /: Merged revisions 80304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80304 | qwell | 2007-08-22 13:25:34 -0500 (Wed, 22 Aug 2007) | 2 - lines Change a syntax that the GNU make in Solaris dislikes. - ........ - - * /, build_tools/make_version: Merged revisions 80302 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r80302 | qwell | 2007-08-22 13:06:00 -0500 (Wed, 22 Aug - 2007) | 3 lines Fix a bashism (we explicitly request /bin/sh). - Remove some oddly placed quotes I found in passing. ........ - -2007-08-22 16:27 +0000 [r80258-80262] Russell Bryant <russell@digium.com> - - * utils/check_expr.c: Ensure that the object code for - ast_atomic_fetchadd_int() gets included in the check_expr binary - when building with LOW_MEMORY defined. (reported by Brian Capouch - on the asterisk-dev list, patch by me) - - * Makefile, /: Merged revisions 80257 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) | - 4 lines Honor the contents of the COPTS variable as custom target - CFLAGS. Apparently this is what openwrt does. (reported by Brian - Capouch on the asterisk-dev list, patch by me) ........ - -2007-08-22 16:16 +0000 [r80256] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 80255 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4 - lines (closes issue #10526) Reported by: sinistermidget Revert - commit from issue #10355 and return timestamp skew to 640. - ........ - -2007-08-22 14:17 +0000 [r80241-80242] Steve Murphy <murf@digium.com> - - * /: blocking 80167 - - * /, main/alaw.c: Merged revisions 80166 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 - line This patch solves problem 1 in 8126; it should not slow down - the alaw codec, but should prevent signal degradation via - multiple trips thru the codec. Fossil estimates the twice thru - this codec will prevent fax from working. 4-6 times thru would - result hearable, noticeable, voice degradation. ........ - -2007-08-21 21:58 +0000 [r80226] Russell Bryant <russell@digium.com> - - * funcs/func_odbc.c: use ast_atomic_fetchadd_int for incrementing - resultcount - -2007-08-21 20:55 +0000 [r80217] Steve Murphy <murf@digium.com> - - * res/ael/pval.c: As per 10472, mvanbaak thought the generated code - would look better this way. - -2007-08-21 18:49 +0000 [r80184] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 80183 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | - 7 lines Don't record SIP dialog history if it's not turned on. - Also, put an upper limit on how many history entires will be - stored for each SIP dialog. It is currently set to 50, but can be - increased if deemed necessary. (closes issue #10421, closes issue - #10418, patches suggested by jmoldenhauer, patches updated by me) - (Security implications documented in AST-2007-020) ........ - -2007-08-21 15:51 +0000 [r80157] Joshua Colp <jcolp@digium.com> - - * main/audiohook.c: Minor tweak. Don't manipulate volume of the - audio in the buffer if no audio is actually there. - -2007-08-21 15:23 +0000 [r80133] Russell Bryant <russell@digium.com> - - * /, channels/chan_mgcp.c: Merged revisions 80132 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) | - 3 lines Don't try to dereference the owner channel when it may - not exist (issue #10507, maxper) ........ - -2007-08-21 15:04 +0000 [r80131] Jason Parker <jparker@digium.com> - - * /, configs/cdr.conf.sample: Merged revisions 80130 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug - 2007) | 7 lines (closes issue #10510) Reported by: casper - Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few - errors in sample cdr config file. ........ - -2007-08-20 22:53 +0000 [r80113] Steve Murphy <murf@digium.com> - - * build_tools/cflags.xml, main/ulaw.c, codecs/slin_ulaw_ex.h, - codecs/ulaw_slin_ex.h, include/asterisk/alaw.h, main/translate.c, - include/asterisk/ulaw.h, main/alaw.c: This change set fixes bug - 8126 in trunk. It is implemented via compile time options, - activated via the menuselect stuff, which defaults to the old - way. non-zero sample data added. Translate tables expressed in - microseconds instead of milliseconds, with 5-digit data now - instead of 3, giving 2 more digits of precision. - -2007-08-20 17:37 +0000 [r80075] Steve Murphy <murf@digium.com> - - * include/asterisk/lock.h, utils/extconf.c: Stephn Davies reports - that this will help make things work on 64-bit machines - -2007-08-20 16:18 +0000 [r80050] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 80049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80049 | mmichelson | 2007-08-20 11:17:43 -0500 (Mon, 20 Aug - 2007) | 4 lines Found a pointless ternary if. member->dynamic was - set to 1 and has no opportunity to change between then and this - line, so "dynamic" will ALWAYS be output. ........ - -2007-08-20 16:12 +0000 [r80048] Jason Parker <jparker@digium.com> - - * /, configs/extensions.conf.sample: Merged revisions 80047 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7 - lines (closes issue #10499) Reported by: casper Patches: - extensions.conf.sample.diff uploaded by casper (license 55) - Update CLI examples in extensions.conf.sample to reflect command - changes. ........ - -2007-08-20 15:53 +0000 [r80046] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Remove remnants of last commit so trunk - builds again. - -2007-08-20 15:37 +0000 [r80045] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 80044 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r80044 | mmichelson | 2007-08-20 10:34:43 -0500 (Mon, 20 Aug - 2007) | 5 lines Ukrainian language voicemail support. (closes - issue #10458, reported and patched by Oleh) ........ - -2007-08-20 15:27 +0000 [r80037] Steve Murphy <murf@digium.com> - - * utils/pval.c (removed): pval.c should not be in svn, in the utils - dir - -2007-08-20 15:10 +0000 [r80023-80033] Joshua Colp <jcolp@digium.com> - - * utils/pval.c: Bring pval.c in utils up to date with pval.c in - res/ael. - - * channels/chan_zap.c: Fix random segfault issue when loading - chan_zap. Trying to access a configuration structure that has - already been destroyed is bad, mmmk? - -2007-08-20 02:46 +0000 [r79999] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 79998 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79998 | tilghman | 2007-08-19 21:42:49 -0500 (Sun, 19 Aug 2007) - | 2 lines Missing curly braces. Oops. (Reported by snuffy via - IRC) ........ - -2007-08-20 00:54 +0000 [r79988-79990] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: (closes issue #10495) Reported by: - stevedavies Make sure context pointer is valid or else chan_iax2 - will go kaboom. - - * utils/Makefile: (closes issue #10496) Reported by: caio1982 Fix - building on OSX. - - * channels/chan_h323.c: Fix building of trunk. I'm doing work on a - Sunday night just to avoid watching Snakes on a Plane which my - roommate is watching. - -2007-08-19 14:17 +0000 [r79980] Tilghman Lesher <tlesher@digium.com> - - * utils/Makefile: Add strcompat dependency for check_expr (needed - for platforms that don't have strndup) - -2007-08-18 23:58 +0000 [r79972] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Actually check the return value of - epoll_create to make sure it works. - -2007-08-18 14:34 +0000 [r79940-79949] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 79947 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79947 | tilghman | 2007-08-18 09:30:44 -0500 (Sat, 18 Aug 2007) - | 3 lines Don't allocate vmu for messagecount when we could just - use the stack instead (closes issue #10490) Also, remove a - useless (and leaky) SQLAllocHandle (closes issue #10480) ........ - - * channels/chan_zap.c, channels/chan_sip.c, channels/chan_h323.c, - channels/chan_iax2.c: We weren't properly encapsulating the mtime - ignores of config files (closes issue #10488) - -2007-08-17 21:19 +0000 [r79915] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: I broke the build. Now I'm fixing it. - -2007-08-17 21:04 +0000 [r79913] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 79912 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) | - 4 lines Avoid a crash in the handling of DTMF based Caller ID. It - is valid for ast_read to return NULL in the case that the channel - has been hung up. (crash reported by anonymouz666 on IRC in - #asterisk-dev) ........ - -2007-08-17 19:16 +0000 [r79907] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 79906 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug - 2007) | 6 lines Patch allows for more seamless transition from - file storage voicemail to ODBC storage voicemail. If a retrieval - of a greeting from the database fails, but the file is found on - the file system, then we go ahead an insert the greeting into the - database. The result of this is that people who switch from file - storage to ODBC storage do not need to rerecord their voicemail - greetings. ........ - -2007-08-17 19:13 +0000 [r79903-79905] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: - Merged revisions 79904 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10430) ........ r79904 | qwell | 2007-08-17 14:12:19 -0500 - (Fri, 17 Aug 2007) | 11 lines Don't send a semicolon over the - wire in sip notify messages. Caused by fix for issue 9938. I - basically took the code that existed before 9938 was fixed, and - copied it into a new function - ast_unescape_semicolon There - should be very few places this will be needed (pbx_config does - NOT need this (see issue 9938 for details)) Issue 10430, patch by - me, with help/ideas from murf (thanks murf). ........ - - * channels/chan_local.c, /: Merged revisions 79902 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500 - (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name - and number. Issue 10485, reported by and fix explained by - paradise. ........ - -2007-08-17 16:39 +0000 [r79901] Tilghman Lesher <tlesher@digium.com> - - * configs/logger.conf.sample: Documentation for %q in logger.conf, - as suggested by jtodd (closes issue #10475) - -2007-08-17 16:04 +0000 [r79888-79894] Jason Parker <jparker@digium.com> - - * res/res_features.c: Fix Dial arguments in res_features. Closes - issue #10484, patch by lunn. - - * pbx/pbx_dundi.c: Correct the argument separator for a Dial - statement in pbx_dundi. Closes issue #10483, patch by lunn - -2007-08-17 14:41 +0000 [r79885] Tilghman Lesher <tlesher@digium.com> - - * main/config.c: Change this flag... might not otherwise unlock in - an OOM situation - -2007-08-17 14:14 +0000 [r79861-79862] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Make use of ast_sched_replace() in some - places in chan_iax2 - - * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: This - commit adds a scheduler API call, ast_sched_replace that can be - used in place of a very common construct. I also used it in a - number of places in chan_sip. if (id > -1) ast_sched_del(sched, - id); id = ast_sched_add(sched, ...); changes to: - ast_sched_replace(id, sched, ...); - -2007-08-17 13:45 +0000 [r79859-79860] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_odbc.c, res/res_config_sqlite.c: store and destroy - implementations for sqlite (closes issue #10446) and odbc (closes - issue #10447) - - * res/res_config_pgsql.c, funcs/func_lock.c: store and destroy - implementations for realtime pgsql (closes issue #10372) - -2007-08-17 13:39 +0000 [r79858] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 79857 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) | - 5 lines Fix some crashes in chan_sip. This patch changes various - places that add items to the scheduler to ensure that they don't - overwrite the ID of a previously scheduled item. If there is one, - it should be removed. (closes issue #10391, closes issue #10256, - probably others, patch by me) ........ - -2007-08-17 08:29 +0000 [r79841] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 79833 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17 - Aug 2007) | 1 line sometimes we don't need to signal dtmf tones - to asterisk, we just want them to go through as inband. Otherwise - they might be generated by the other channel partner and then - there is a double tone. ........ - -2007-08-17 01:19 +0000 [r79824] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Fix building of chan_zap under development - mode without libpri and libss7 installed. - -2007-08-16 23:31 +0000 [r79813] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_lock.c: Revise dialplan locks to permit multiple locks - per channel, but with deadlock avoidance - -2007-08-16 22:33 +0000 [r79764-79794] Russell Bryant <russell@digium.com> - - * /: Merged revisions 79792 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79792 | russell | 2007-08-16 17:32:33 -0500 (Thu, 16 Aug 2007) | - 4 lines Fix a little race condition that could cause a crash if - two channels had MOH stopped at the same time that were using a - class that had been marked for deletion when its use count hits - zero. ........ - - * /, res/res_musiconhold.c: Merged revisions 79778 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16 - Aug 2007) | 14 lines This patch fixes a bug where reloading the - module with "module reload" did not delete classes from memory - that were no longer in the config. This patch fixes that problem - as well as another one. Previously, if you reloaded MOH using the - "moh reload" CLI command, which behaved differently than "module - reload ...", MOH had to be stopped on every channel and started - again immediately. However, there was no way to tell what class - was being used, so they would all fall back to the default class. - (closes issue #10139) Reported by: blitzrage Patches: - asterisk-10139-advanced.diff.txt uploaded by jamesgolovich - (license 176) Tested by: jamesgolovich ........ - - * /, channels/chan_iax2.c: Merged revisions 79756 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79756 | russell | 2007-08-16 16:29:24 -0500 (Thu, 16 Aug 2007) | - 11 lines Fix more deadlocks in chan_iax2 that were introduced by - making frame handling and scheduling multi-threaded. - Unfortunately, we have to do some expensive deadlock avoidance - when queueing frames on to the ast_channel owner of the IAX2 pvt - struct. This was already handled for regular frames, but - ast_queue_hangup and ast_queue_control were still used directly. - Making these changes introduced even more places where the IAX2 - pvt struct can disappear in the context of a function holding its - lock due to calling a function that has to unlock/lock it to - avoid deadlocks. I went through and fixed all of these places to - account for this possibility. (issue #10362, patch by me) - ........ - -2007-08-16 21:28 +0000 [r79755] Joshua Colp <jcolp@digium.com> - - * /: Fix properties on trunk again. - -2007-08-16 21:21 +0000 [r79749] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 79748 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r79748 | mmichelson | 2007-08-16 16:16:40 -0500 (Thu, 16 - Aug 2007) | 8 lines Fixes a problem where agents would get stuck - busy due to their wrapuptime being longer than the queue's - wrapuptime and ringinuse=no for the queue. (closes issue #10215, - reported by Doug, repaired by me) Special thanks to fkasumovic - for pointing out the source of the problem and to bweschke for - helping to come up with a solution! ........ - -2007-08-16 21:09 +0000 [r79747] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, cdr/cdr_sqlite3_custom.c, /, res/res_features.c, - codecs/codec_adpcm.c, apps/app_alarmreceiver.c, - cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/config.c, - main/loader.c, res/res_smdi.c, channels/chan_skinny.c, - main/http.c, apps/app_amd.c, channels/chan_alsa.c, - cdr/cdr_odbc.c, cdr/cdr_manager.c, codecs/codec_g722.c, - apps/app_privacy.c, codecs/codec_speex.c, channels/chan_agent.c, - codecs/codec_g726.c, channels/iax2-provision.c, - apps/app_playback.c, channels/iax2-provision.h, - channels/chan_misdn.c, res/res_indications.c, pbx/pbx_config.c, - main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, - channels/chan_vpb.cc, res/res_snmp.c, apps/app_meetme.c, - codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c, - cdr/cdr_pgsql.c, apps/app_followme.c, res/res_jabber.c, - cdr/cdr_radius.c, codecs/codec_zap.c, res/res_config_sqlite.c, - main/enum.c, channels/misdn_config.c, cdr/cdr_csv.c, main/cdr.c, - channels/chan_phone.c, res/res_config_odbc.c, main/manager.c, - apps/app_osplookup.c, funcs/func_odbc.c, apps/app_minivm.c, - main/logger.c, apps/app_directory.c, apps/app_rpt.c, - cdr/cdr_custom.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, - res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, - channels/chan_sip.c, apps/app_festival.c, codecs/codec_alaw.c, - res/res_adsi.c, include/asterisk/config.h, apps/app_queue.c, - channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c, - channels/chan_jingle.c, channels/misdn/chan_misdn_config.h, - channels/chan_h323.c, pbx/pbx_dundi.c, codecs/codec_ulaw.c: Don't - reload a configuration file if nothing has changed. - -2007-08-16 19:40 +0000 [r79736] Steve Murphy <murf@digium.com> - - * utils/pval.c, utils/conf2ael.c: Many thanks to mvanbaak for his - update to translate hints; I added the -d option for local - testing purposes. This is from bug 10472 - -2007-08-16 18:23 +0000 [r79724-79725] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_iax2.c: added counter for iax2 show registry CLI - output, closes issue 10461, thanks junky - - * apps/app_voicemail.c: added counter for voicemail show users, - issue 10462, thanks junky - -2007-08-16 17:34 +0000 [r79714-79719] Steve Murphy <murf@digium.com> - - * utils/conf2ael.c: mvanbaak asks: why did you include that twice? - Answer: dunno. removed redundant include - - * utils/extconf.c, utils/conf2ael.c: svn did me dirty for some - reason. Left 5 files out of the commit; Tilghman copied them in - from the branch, but I had made changes to these. Here they are. - -2007-08-16 15:59 +0000 [r79691] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 79690 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79690 | mmichelson | 2007-08-16 10:58:34 -0500 (Thu, 16 Aug - 2007) | 5 lines base_encode is not trying to open a log file, so - we should not call it a log file in the warning. (related to - issue #10452, reported by bcnit) ........ - -2007-08-16 15:29 +0000 [r79687-79688] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_dundi.c: (closes issue #10467) Reported by: lunn Patches: - pbx_dundi.diff uploaded by lunn (license 179) Don't print a - warning saying an ethernet interface was found when it indeed - was. - - * utils/conf2ael.c: Make conf2ael build on 64-bit systems. - -2007-08-16 09:45 +0000 [r79666] Philippe Sultan <philippe.sultan@gmail.com> - - * /, res/res_jabber.c: Merged revisions 79665 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007) - | 21 lines A fix for two critical problems detected while working - with Daniel McKeehan in issue #10184. Upon priority change, the - resource list is not NULL terminated when moving an item to the - end of the list. This makes Asterisk endlessy loop whenever it - needs to read the list. Jids with different resource and priority - values, like in Gmail's and GoogleTalk's jabber clients put that - problem in evidence. Upon reception of a 'from' attribute with an - empty resource string, Asterisk crashes when trying to access the - found->cap pointer if the resource list for the given buddy is - not empty. This situation is perfectly valid and must be handled. - The Gizmoproject's jabber client put that problem in evidence. - Also added a few comments in the code as well as a handle for the - capabilities from Gmail's jabber client, which are stored in a - caps:c tag rather than the usual c tag. Closes issue #10184. - ........ - -2007-08-16 09:22 +0000 [r79660] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/ie.c: Merged revisions 79642 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79642 | crichter | 2007-08-16 10:21:21 +0200 (Do, 16 Aug 2007) | - 1 line 0x80 + protocol is wrong for USERUSER when we want to send - IA5 Chars. ........ - -2007-08-16 06:52 +0000 [r79638] Olle Johansson <oej@edvina.net> - - * CHANGES: Doc change - -2007-08-15 22:53 +0000 [r79634] Jason Parker <jparker@digium.com> - - * res/res_musiconhold.c: Modify the names of functions/variables in - res_musiconhold to be useful. Closes issue #10464, patch by - caio1982 - -2007-08-15 21:25 +0000 [r79623] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/pval.h (added), utils/pval.c (added), - include/asterisk/extconf.h (added), utils/extconf.c (added), - utils/conf2ael.c (added): Missing from murf's last trunk commit, - which was why trunk won't compile - -2007-08-15 19:34 +0000 [r79611] Joshua Colp <jcolp@digium.com> - - * /: Remove properties that appeared from Steve's last branch - merge. Automerge has already run so everyone's branches based off - of trunk are probably toast by now. - -2007-08-15 19:21 +0000 [r79595] Steve Murphy <murf@digium.com> - - * /, pbx/ael/ael.y (removed), pbx/ael/ael-test/ref.ael-test11, - res/Makefile, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test16, - pbx/ael/ael-test/ref.ael-test19, include/asterisk/ast_expr.h, - pbx/ael/ael_lex.c (removed), pbx/pbx_ael.c, pbx/ael/ael.flex - (removed), res/ael (added), main/pbx.c, UPGRADE.txt, - res/res_ael_share.c (added), pbx/Makefile, CHANGES, - utils/Makefile, pbx/ael/ael-test/ref.ael-ntest10, - pbx/ael/ael.tab.c (removed), pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/ael/ael-test/ref.ael-test4, include/asterisk/ael_structs.h, - pbx/ael/ael.tab.h (removed), pbx/ael/ael-test/ref.ael-test5, - utils/ael_main.c, include/asterisk/pbx.h, - pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, - utils/check_expr.c: This commit closes bug 7605, and half-closes - 7638. The AEL code has been redistributed/repartitioned to allow - code re-use both inside and outside of Asterisk. This commit - introduces the utils/conf2ael program, and an external - config-file reader, for both normal config files, and for - extensions.conf (context, exten, prio); It provides an API for - programs outside of asterisk to use to play with the dialplan and - config files. - -2007-08-15 14:42 +0000 [r79558] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 79553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 - lines (closes issue #10440) Reported by: irroot (closes issue - #10454) Reported by: flo_turc Increase maximum timestamp skew to - 120. 20 was apparently far too low. ........ - -2007-08-15 14:27 +0000 [r79529] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 79527 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug - 2007) | 5 lines Fixed an error in the Russian language voicemail - intro. (issue #10458, reported and patched by Oleh) ........ - -2007-08-15 14:20 +0000 [r79524] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 79523 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 - lines (closes issue #10456) Reported by: irroot Patches: - sip_timeout.patch uploaded by irroot (license 52) Change - hardcoded timer value to defined value. I'm doing this in 1.4 as - well so if it needs to be changed in the future this place would - not have been forgotten. ........ - -2007-08-15 11:27 +0000 [r79507] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: Merged revisions 78936 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) | - 1 line fixed a bug with the useruser information element. We send - them now also in the disconnect message. ........ - -2007-08-14 18:50 +0000 [r79437-79471] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 79470 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) | - 2 lines Fix another spot where an iax2_peer would be leaked if - realtime was in use. ........ - - * /, channels/chan_iax2.c: Merged revisions 79436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79436 | russell | 2007-08-14 12:31:39 -0500 (Tue, 14 Aug 2007) | - 3 lines Fix some memory leaks throughout chan_iax2 related to the - use of realtime. I found these while working on iax2_peer object - reference tracking. ........ - -2007-08-14 15:30 +0000 [r79403] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 79397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4 - lines (closes issue #10415) Reported by: atis Revert fix for - #10327 as it causes more issues then it solves. ........ - -2007-08-14 14:32 +0000 [r79392] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest17, /, - pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ael-test5/extensions.ael, - pbx/ael/ael-test/ael-test6/extensions.ael, - pbx/ael/ael-test/ref.ael-test19, - pbx/ael/ael-test/ael-vtest21/extensions.ael, - pbx/ael/ael-test/ael-vtest21 (added), - pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test2, pbx/pbx_ael.c, - pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, - utils/ael_main.c, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-vtest21 (added), - pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 79255 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79255 | murf | 2007-08-13 11:49:54 -0600 (Mon, 13 Aug 2007) | 1 - line This patch fixes bug 10411. I added a new regression test, - some regression test cleanups ........ - -2007-08-14 14:17 +0000 [r79379] Joshua Colp <jcolp@digium.com> - - * main/channel.c: (closes issue #10427) Reported by: pj Two of the - three places ast_waitfor_nandfds could branch off to did not - clear outfd and exception. If the calling function did not clear - these there was a chance they could get a false positive on - testing to see whether they were set. - -2007-08-14 13:46 +0000 [r79378] Steve Murphy <murf@digium.com> - - * main/channel.c, channels/chan_zap.c: Don't ask me why, but - waitfordigit will immediately return a 1 on my system, unless the - outfd is initialized to -1 before calling the nandfds func - -2007-08-13 21:59 +0000 [r79335] Joshua Colp <jcolp@digium.com> - - * /, include/asterisk/speech.h, res/res_speech.c, - apps/app_speech_utils.c: Merged revisions 79334 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2 - lines Instead of accepting a single DTMF character accept a full - string. ........ - -2007-08-13 21:44 +0000 [r79333] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c: Only use the sanitysql if it's not zero-len - -2007-08-13 20:40 +0000 [r79273-79306] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 79301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79301 | russell | 2007-08-13 15:37:50 -0500 (Mon, 13 Aug 2007) | - 3 lines Don't call find_peer in registry_authrequest with the pvt - lock held to avoid a deadlock. ........ - - * /, channels/chan_iax2.c: Merged revisions 79276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79276 | russell | 2007-08-13 15:18:30 -0500 (Mon, 13 Aug 2007) | - 4 lines Release the pvt lock before calling find_peer in - register_verify to avoid a deadlock. Also, remove some - unnecessary locking in auth_fail that was only done recursively. - ........ - - * /, channels/chan_iax2.c: Merged revisions 79274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79274 | russell | 2007-08-13 15:02:57 -0500 (Mon, 13 Aug 2007) | - 3 lines Don't call find_peer within update_registry with a pvt - lock held. This can cause a deadlock as the code will eventually - call find_callno. ........ - - * /, channels/chan_iax2.c: Merged revisions 79272 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79272 | russell | 2007-08-13 14:27:39 -0500 (Mon, 13 Aug 2007) | - 9 lines I am fighting deadlocks in chan_iax2. I have tracked them - down to a single core issue. You can not call find_callno() while - holding a pvt lock as this function has to lock another (every) - other pvt lock. Doing so can lead to a classic deadlock. So, I am - tracking down all of the code paths where this can happen and - fixing them. The fix I committed earlier today was along the same - theme. This patch fixes some code down the path of - authenticate_reply. ........ - -2007-08-13 15:39 +0000 [r79238] Mark Michelson <mmichelson@digium.com> - - * CHANGES, apps/app_queue.c: Allow non-realtime queues to have - realtime members (issue #10424, reported and patched by irroot) - -2007-08-13 15:32 +0000 [r79222] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 79214 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79214 | russell | 2007-08-13 10:28:13 -0500 (Mon, 13 Aug 2007) | - 4 lines Fix a potential deadlock in socket_process. - check_provisioning can eventually call find_callno. You can't - hold a pvt lock while calling find_callno because it goes through - and locks every single one looking for a match. ........ - -2007-08-13 14:55 +0000 [r79208] Joshua Colp <jcolp@digium.com> - - * /, include/asterisk/speech.h, res/res_speech.c, - apps/app_speech_utils.c: Merged revisions 79207 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2 - lines Add an API call to allow the engine to know that DTMF was - received. ........ - -2007-08-13 14:23 +0000 [r79176] Russell Bryant <russell@digium.com> - - * main/channel.c, include/asterisk/channel.h: constify the return - value of reason2str - -2007-08-13 14:22 +0000 [r79175] Joshua Colp <jcolp@digium.com> - - * channels/chan_jingle.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_zap.c, /, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, - channels/chan_mgcp.c: Merged revisions 79174 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 - lines (closes issue #10437) Reported by: haklin Don't set the - callerid name and number a second time on a newly created - channel. ast_channel_alloc itself already sets it and setting it - twice would cause a memory leak. ........ - -2007-08-11 05:28 +0000 [r79147] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 79142 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79142 | tilghman | 2007-08-11 00:23:04 -0500 (Sat, 11 Aug 2007) - | 2 lines Ensure the connection gets marked as used at allocation - time (closes issue #10429, report and fix by mnicholson) ........ - -2007-08-10 21:29 +0000 [r79109] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Use localized softkey labels. Add some - information about localization "codes". - -2007-08-10 21:03 +0000 [r79100] Steve Murphy <murf@digium.com> - - * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: - Merged revisions 79099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1 - line From a user complaint on #asterisk, I have forced pbx_spool - to explain what reason codes mean, when they are logged ........ - -2007-08-10 20:48 +0000 [r79098] Russell Bryant <russell@digium.com> - - * funcs/func_devstate.c: Store custom device states in astdb so - that they will persist a restart. As a side benefit, this - simplifies the code a bit, too. - -2007-08-10 18:37 +0000 [r79074] Joshua Colp <jcolp@digium.com> - - * main/dial.c: Bring up to date with poll changes. - -2007-08-10 18:35 +0000 [r79045-79068] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 79049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79049 | murf | 2007-08-10 12:25:51 -0600 (Fri, 10 Aug 2007) | 1 - line Re bug behavior mentioned in #asterisk, made this tweak to - code, to prevent hundreds of log messages from being generated - ........ - - * /: oops. forgot to commit the prop change on . - - * main/cdr.c: Merged revisions 79044 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r79044 | murf | 2007-08-10 11:43:49 -0600 (Fri, 10 Aug 2007) | 1 - line This will help debug; from a question asked on #asterisk - ........ - -2007-08-10 16:24 +0000 [r79005-79027] Russell Bryant <russell@digium.com> - - * include/asterisk/devicestate.h, apps/app_meetme.c, - res/res_features.c, main/devicestate.c, main/event.c, - funcs/func_devstate.c: Merge a set of device state improvements - from team/russell/events. The way a device state change - propagates is kind of silly, in my opinion. A device state - provider calls a function that indicates that the state of a - device has changed. Then, another thread goes back and calls a - callback for the device state provider to find out what the new - state is before it can go send it off to whoever cares. I have - changed it so that you can include the state that the device has - changed to in the first function call from the device state - provider. This removes the need to have to call the callback, - which locks up critical containers to go find out what the state - changed to. This change set changes the "simple" device state - providers to use the new method. This includes parking, meetme, - and SLA. I have also mostly converted chan_agent in my branch, - but still have some more things to think through before - presenting the plan for converting channel drivers to ensure all - of the right events get generated ... - - * /, include/asterisk/lock.h: Merged revisions 78995 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10 - Aug 2007) | 4 lines The last set of changes that I made to "core - show locks" made it not able to track mutexes unless they were - declared using AST_MUTEX_DEFINE_STATIC. Locks initialized with - ast_mutex_init() were not tracked. It should work now. ........ - -2007-08-10 14:17 +0000 [r78952-78956] Joshua Colp <jcolp@digium.com> - - * /, main/file.c: Merged revisions 78955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2 - lines Don't bother having the core pass through or emulate begin - DTMF frames when in an ast_waitstream. It only cares about the - end of DTMF. ........ - - * /, configs/queues.conf.sample: Merged revisions 78951 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78951 | file | 2007-08-10 10:49:19 -0300 (Fri, 10 Aug 2007) | 4 - lines (closes issue #10422) Reported by: bhowell Add note to - sample configuration about module load order and how it can cause - perfectly good queue members to be marked as invalid. ........ - -2007-08-09 23:49 +0000 [r78908] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 78907 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78907 | mmichelson | 2007-08-09 18:47:00 -0500 (Thu, 09 Aug - 2007) | 4 lines Improved a bit of logic regarding comma-separated - mailboxes in has_voicemail. Also added some braces to some - compound if statements since unbraced if statements scare me in - general. ........ - -2007-08-09 23:32 +0000 [r78906] Steve Murphy <murf@digium.com> - - * Makefile, /: Merged revisions 78891 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78891 | murf | 2007-08-09 17:10:46 -0600 (Thu, 09 Aug 2007) | 1 - line This fixes bug 10416; thanks to mvanbaak for the pretty - output ........ - -2007-08-09 22:19 +0000 [r78861-78862] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 78859 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78859 | mmichelson | 2007-08-09 16:51:17 -0500 (Thu, 09 Aug - 2007) | 9 lines Quite a few changes regarding IMAP storage. 1. - instead of using inboxcount as the core message counting - function, we use messagecount instead. This makes it possible to - count messages in folders besides just INBOX and Old. 2. - inboxcount and hasvoicemail now use messagecount as their means - of determining return values. 3. Added a copy_message function - for IMAP storage. Unfortunately I don't have the means to test - it, but it seems like a pretty straightforward function. 4. - Removed a #ifndef IMAP_STORAGE and matching #endif from - leave_voicemail for a couple of reasons. One, we want to support - copying mail to multiple IMAP boxes, and two, IMAP was broken - because a STORE macro had been moved into this section of code. - ........ - -2007-08-09 20:07 +0000 [r78829] Russell Bryant <russell@digium.com> - - * apps/app_minivm.c: Don't use strncpy for moving a chunk of memory - to another that is overlapping. This was found by running - Asterisk under valgrind. - -2007-08-09 19:35 +0000 [r78718-78824] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: When looking up a mailbox, use the default - context if not specified as something else - - * channels/chan_sip.c: Restore the ability to have multiple - mailboxes listed for the mailbox option in sip.conf. chan_sip now - maintains separate internal MWI subscriptions for each one. - - * /, apps/app_voicemail.c: Merged revisions 78778 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78778 | russell | 2007-08-09 12:58:31 -0500 (Thu, 09 Aug 2007) | - 1 line add a comment to indicate that inboxcount for ODBC_STORAGE - needs to be fixed to support multiple mailboxes ........ - - * /, apps/app_voicemail.c: Merged revisions 78749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78749 | russell | 2007-08-09 12:24:40 -0500 (Thu, 09 Aug 2007) | - 9 lines Fix subscriptions to multiple mailboxes for ODBC_STORAGE. - Also, leave a comment for this to be fixed for IMAP_STORAGE, as - well. I left IMAP alone since I know MarkM was working on this - code right now for another reason. This is broken even worse in - trunk, but for a different reason. The fact that the mailbox - option supported multiple mailboxes is completely not obvious - from the code in the channel drivers. Anyway, I will fix that in - another commit ... ........ - - * channels/chan_zap.c, channels/chan_sip.c, - include/asterisk/event_defs.h, channels/chan_iax2.c, - channels/chan_mgcp.c, apps/app_voicemail.c: Fix a problem that I - had introduced into MWI handling. I had ignored the mailbox - context. Now, all related MWI event dealings pay attention to the - context as well. - - * /, apps/app_meetme.c: Merged revisions 78717 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) | - 7 lines Fix a problem with the combination of the 'F' option to - pass DTMF through a conference and options that use DTMF to - activate various features. The problem was that the BEGIN frame - would be passed through, but the END frame would get intercepted - to activate a feature. Then, the other conference members would - hear DTMF for forever, which they didn't seem to like very much. - (closes issue #10400, reported by stevefeinstein, fixed by me) - ........ - -2007-08-08 22:05 +0000 [r78649-78686] Joshua Colp <jcolp@digium.com> - - * configure: Regenerate configure script. This actually just - updated the revision number... since my last merge changed it to - an older number, while it was in fact generated from a much newer - revision. - - * channels/chan_skinny.c: Minor fix for building under dev mode - when byteswapping macro header files are not available. - - * apps/app_dial.c, channels/chan_zap.c, channels/chan_sip.c, - include/asterisk/autoconfig.h.in, channels/chan_agent.c, - configure.ac, include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_oss.c, main/rtp.c, main/channel.c, - channels/chan_jingle.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, configure, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c: - Add support for using epoll instead of poll. This should increase - scalability and is done in such a way that we should be able to - add support for other poll() replacements. - - * channels/chan_zap.c: HAVEL_SS7 should be HAVE_SS7. Reported by - kwallace. - - * main/channel.c, include/asterisk/audiohook.h (added), - funcs/func_volume.c (added), main/Makefile, main/slinfactory.c, - include/asterisk/chanspy.h (removed), include/asterisk/channel.h, - main/audiohook.c (added), apps/app_chanspy.c, - apps/app_mixmonitor.c, include/asterisk/slinfactory.h: Merge - audiohooks branch into trunk. This is a new API for developers to - listen and manipulate the audio going through a channel. - -2007-08-08 19:30 +0000 [r78648] Jason Parker <jparker@digium.com> - - * /, doc/jabber.txt: Merged revisions 78646 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78646 | qwell | 2007-08-08 14:29:42 -0500 (Wed, 08 Aug 2007) | 2 - lines Fix mogs email address. ........ - -2007-08-08 19:03 +0000 [r78637] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Correct spelling. s/threaads/threads/ - -2007-08-08 18:34 +0000 [r78590-78635] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 78575 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78575 | mmichelson | 2007-08-08 09:26:36 -0500 (Wed, 08 Aug - 2007) | 4 lines Changing a bit of logic so that someone will - NEVER exit the queue on timeout unless they have enabled the 'n' - option. This commit relates to issue #10320. Thanks to - jfitzgibbon for detailing the idea behind this code change. - ........ - -2007-08-08 13:52 +0000 [r78570] Joshua Colp <jcolp@digium.com> - - * /, configs/sip.conf.sample: Merged revisions 78569 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug - 2007) | 4 lines (closes issue #10335) Reported by: adamgundy - Update sip.conf to include another scenario where directrtpsetup - will fail. ........ - -2007-08-07 23:04 +0000 [r78541] Russell Bryant <russell@digium.com> - - * main/pbx.c, pbx/pbx_spool.c, main/sha1.c, res/res_features.c, - res/res_crypto.c, utils/smsq.c, include/asterisk/features.h: Add - another big set of doxygen documentation improvements from - snuffy. (closes issue #9892) (closes issue #10395) - -2007-08-07 22:13 +0000 [r78521] Joshua Colp <jcolp@digium.com> - - * main/manager.c, include/asterisk/manager.h: Use the linkedlists.h - macros for the manager action list. - -2007-08-07 21:00 +0000 [r78489] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 78488 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78488 | russell | 2007-08-07 15:57:54 -0500 (Tue, 07 - Aug 2007) | 2 lines Fix the build of this module on 64-bit - platforms ........ - -2007-08-07 19:44 +0000 [r78451] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 78450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78450 | mmichelson | 2007-08-07 14:43:57 -0500 (Tue, 07 Aug - 2007) | 5 lines The logic behind inboxcount's return value was - reversed in has_voicemail and message_count. (closes issue - #10401, reported by st1710, patched by me) ........ - -2007-08-07 19:36 +0000 [r78442] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 78437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78437 | tilghman | 2007-08-07 14:34:25 -0500 (Tue, 07 Aug 2007) - | 2 lines Don't free the environment handle when the connection - fails, because other connections might be depending upon it - ........ - -2007-08-07 19:14 +0000 [r78417] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_odbc.c, /, apps/app_directory.c, - apps/app_voicemail.c: Merged revisions 78415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78415 | tilghman | 2007-08-07 14:09:38 -0500 (Tue, 07 Aug 2007) - | 2 lines Reconnection doesn't happen automatically when a DB - goes down (fixes issue #9389) ........ - -2007-08-07 18:26 +0000 [r78378] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 78375 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78375 | qwell | 2007-08-07 13:25:15 -0500 (Tue, 07 Aug - 2007) | 3 lines Properly check the capabilities count to avoid a - segfault. (ASA-2007-019) ........ - -2007-08-07 17:46 +0000 [r78372] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 78371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r78371 | russell | 2007-08-07 12:45:30 -0500 - (Tue, 07 Aug 2007) | 12 lines Merged revisions 78370 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 - Aug 2007) | 4 lines Revert patch committed for issue #9660. It - broke E&M trunks. (closes issue #10360) (closes issue #10364) - ........ ................ - -2007-08-07 16:17 +0000 [r78346-78347] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Can't forget outsignaling! - - * channels/chan_zap.c: Just for jsmith... make signaling a valid - option that acts like signalling. - -2007-08-07 16:04 +0000 [r78342] Russell Bryant <russell@digium.com> - - * res/res_eventtest.c (removed): Remove some test code from trunk - as it doesn't need to be here. I'm just going to keep it with a - bunch of other changes i have sitting in a branch. - -2007-08-07 15:40 +0000 [r78338] Joshua Colp <jcolp@digium.com> - - * main/frame.c: (closes issue #10225) Reported by: klaus3000 Clean - up AST_FORMAT_LIST list. It may have mattered in the old days to - have undefined entries but these days it does not. - -2007-08-06 23:00 +0000 [r78312] Jason Parker <jparker@digium.com> - - * channels/chan_agent.c: Add a TalkingToChan to the response of the - "agents" manager action. This is similar to the existing "talking - to" that you see what using the "agent show" CLI command. Closes - issue #10102 - -2007-08-06 21:59 +0000 [r78276-78279] Joshua Colp <jcolp@digium.com> - - * apps/app_senddtmf.c: Fix bug where a NULL timeout would make - things explode if SendDTMF was called with it. - - * apps/app_dial.c, main/channel.c, include/asterisk/app.h, - res/res_features.c, apps/app_test.c, main/app.c, - include/asterisk/channel.h, apps/app_senddtmf.c: Extend the - ast_senddigit and ast_dtmf_stream API calls to allow the duration - of the DTMF digit(s) to be specified and make the SendDTMF - application have the capability to use it. - - * main/channel.c, /: Merged revisions 78275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2 - lines Add additional DTMF log messages to help when debugging - issues. ........ - -2007-08-06 20:45 +0000 [r78243] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 78242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78242 | russell | 2007-08-06 15:44:09 -0500 (Mon, 06 Aug 2007) | - 4 lines Fix an issue where dynamic threads can get free'd, but - still exist in the dynamic thread list. (closes issue #10392, - patch from Mihai, with credit to his colleague, Pete) ........ - -2007-08-06 19:52 +0000 [r78227] Doug Bailey <dbailey@digium.com> - - * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c, - main/fskmodem.c: Change the fsk filter used in CID and TDD decode - to an integer based implementation - -2007-08-06 17:51 +0000 [r78186-78192] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fixing a compiler warning which warns that a - variable may be used unitialized. Thanks to mvanbaak for pointing - this out. - - * /, channels/chan_sip.c, include/asterisk/config.h, main/config.c: - Merged revisions 78103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug - 2007) | 7 lines Changed the behavior of sip's realtime_peer - function to match the corresponding way of matching for - non-realtime peers. Now matches are made on both the IP address - and port number, or if the insecure setting is set to "port" then - just match on the IP address. In order to accomplish this, I also - added a new API call, ast_category_root, which returns the first - variable of an ast_category struct ........ - -2007-08-06 16:51 +0000 [r78185] Russell Bryant <russell@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 78184 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) | - 5 lines Fix the return value of AST_LIST_REMOVE(). This shouldn't - be causing any problems, though, because the only code that uses - the return value only checks to see if it is NULL. (closes issue - #10390, pointed out by mihai) ........ - -2007-08-06 16:34 +0000 [r78183] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 78182 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78182 | file | 2007-08-06 13:32:44 -0300 (Mon, 06 Aug 2007) | 2 - lines It is possible for a transfer to occur before the remote - device has our tag in which case they send none in the transfer. - In this case we need to not fail the transfer dialog lookup. - ........ - -2007-08-06 16:31 +0000 [r78179-78181] Jason Parker <jparker@digium.com> - - * /, main/config.c: Merged revisions 78180 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #9938) ........ r78180 | qwell | 2007-08-06 11:30:51 -0500 - (Mon, 06 Aug 2007) | 5 lines Fix an issue with using UpdateConfig - (manager action) where escaped semicolons in a config would be - converted to just semicolons (\; to ;) Issue 9938 ........ - - * channels/chan_skinny.c, configs/skinny.conf.sample: Implement - setvar functionality in chan_skinny Closes issue #10379, patch by - mvanbaak. - -2007-08-06 15:28 +0000 [r78167-78173] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 78172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 - lines (closes issue #10355) Reported by: wdecarne Now that we - pass through RTP timestamp information we need to make the - allowed timestamp skew considerably less. There are situations - where a source may change and due to the timestamp difference the - receiver will experience an audio gap since we did not indicate - by setting the marker bit that the source changed. ........ - - * apps/app_externalivr.c: (closes issue #10381) Reported by: yehavi - Use the filename we parsed using the standard parsing when - launching the application specified to ExternalIVR. - - * /, configure, configure.ac: Merged revisions 78166 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78166 | file | 2007-08-06 11:18:20 -0300 (Mon, 06 Aug - 2007) | 4 lines (closes issue #10383) Reported by: rizzo Include - stdlib.h so NULL gets defined for gethostbyname_r checks. - ........ - -2007-08-05 04:16 +0000 [r78142-78144] Russell Bryant <russell@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 78143 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r78143 | russell | 2007-08-04 23:15:31 -0500 (Sat, 04 - Aug 2007) | 2 lines Fix compilation failure when MALLOC_DEBUG is - enabled, but DEBUG_THREADS is not ........ - - * apps/app_exec.c: Make this module build on my mac - -2007-08-05 03:42 +0000 [r78140] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 78139 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78139 | tilghman | 2007-08-04 22:29:01 -0500 (Sat, 04 Aug 2007) - | 2 lines If peer is not found, the error message is misleading - (should be peer not found, not ACL failure) ........ - -2007-08-05 03:14 +0000 [r78138] Russell Bryant <russell@digium.com> - - * include/asterisk/linkedlists.h: Fix building res_crypto on - systems that init locks with constructors. The problem was that - res_crypto now has a RWLIST named "keys". The macro for defining - this list defines a function used as a constructor for the list - called "init_keys". However, there was another function called - init_keys in this module for a CLI command. The fix is just to - prepend the generated functions with underscores. - -2007-08-03 20:21 +0000 [r78029-78102] Russell Bryant <russell@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 78101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78101 | russell | 2007-08-03 15:14:06 -0500 (Fri, 03 Aug 2007) | - 10 lines (closes issue #10194) Reported by: blitzrage Patches: - bug0010194 uploaded by vovochka Tested by: blitzrage Fix a - problem when you call Voicemail() with multiple mailboxes - specified and ODBC_STORAGE is in use. The audio part of the - message was only given to the first mailbox specified. ........ - - * /, main/utils.c, include/asterisk/lock.h, main/astmm.c: Merged - revisions 78095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) | - 28 lines Add some improvements to lock debugging. These changes - take effect with DEBUG_THREADS enabled and provide the following: - * This will keep track of which locks are held by which thread as - well as which lock a thread is waiting for in a thread-local data - structure. A reference to this structure is available on the - stack in the dummy_start() function, which is the common entry - point for all threads. This information can be easily retrieved - using gdb if you switch to the dummy_start() stack frame of any - thread and print the contents of the lock_info variable. * All of - the thread-local structures for keeping track of this lock - information are also stored in a list so that the information can - be dumped to the CLI using the "core show locks" CLI command. - This introduces a little bit of a performance hit as it requires - additional underlying locking operations inside of every - lock/unlock on an ast_mutex. However, the benefits of having this - information available at the CLI is huge, especially considering - this is only done in DEBUG_THREADS mode. It means that in most - cases where we debug deadlocks, we no longer have to request - access to the machine to analyze the contents of ast_mutex_t - structures. We can now just ask them to get the output of "core - show locks", which gives us all of the information we needed in - most cases. I also had to make some additional changes to astmm.c - to make this work when both MALLOC_DEBUG and DEBUG_THREADS are - enabled. I disabled tracking of one of the locks in astmm.c - because it gets used inside the replacement memory allocation - routines, and the lock tracking code allocates memory. This - caused infinite recursion. ........ - - * /, channels/chan_iax2.c: Merged revisions 78063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78063 | russell | 2007-08-03 12:01:07 -0500 (Fri, 03 Aug 2007) | - 4 lines Only pass through HOLD and UNHOLD control frames when the - mohinterpret option is set to "passthrough". This was pointed out - by Kevin in the middle of a training session. ........ - - * /, channels/chan_iax2.c: Merged revisions 78028 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r78028 | russell | 2007-08-02 21:04:22 -0500 (Thu, 02 Aug 2007) | - 6 lines Don't reuse the timespec that was set to 0 in the - previous timedwait as it will just return immediately. Also, fix - some logic so the thread's lock isn't unlocked twice in the weird - case of dynamic threads getting acquired right after a timeout. - (pointed out by SteveK) ........ - -2007-08-02 21:54 +0000 [r77994-77997] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged - revisions 77996 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #9779) ........ r77996 | qwell | 2007-08-02 16:53:39 -0500 - (Thu, 02 Aug 2007) | 5 lines Make sure we actually allow 6 chars - to be sent. Also make note of the "A" option of date format. - Issue 9779, modifications by DEA, wedhorn, and myself. ........ - - * /, channels/chan_skinny.c: Merged revisions 77993 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10325) ........ r77993 | qwell | 2007-08-02 15:22:40 -0500 - (Thu, 02 Aug 2007) | 5 lines If a device disconnects, the session - will go away. If this happens during call setup, we need to give - up. Issue 10325. ........ - -2007-08-02 19:26 +0000 [r77950] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 77949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77949 | russell | 2007-08-02 14:25:14 -0500 (Thu, 02 Aug 2007) | - 5 lines Fix the case where a dynamic thread times out waiting for - something to do during the first time it runs. This shouldn't - ever happen, but we should account for it anyway. (pointed out by - pete, who works with mihai) ........ - -2007-08-02 18:43 +0000 [r77948] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 77947 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10299) ........ r77947 | qwell | 2007-08-02 13:42:36 -0500 - (Thu, 02 Aug 2007) | 5 lines Make sure we clear the prompt status - message on a hangup. Also rearrange messages to better fit with - what a wireshark trace shows it should be. Issue 10299, initial - patch and solution by sbisker, modified by me to fit with - wireshark trace. ........ - -2007-08-02 18:32 +0000 [r77946] Steve Murphy <murf@digium.com> - - * /, main/fskmodem.c: Merged revisions 77945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r77945 | murf | 2007-08-02 12:21:40 -0600 (Thu, - 02 Aug 2007) | 9 lines Merged revisions 77942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 - line This patch hopefully solves 10141; The user is running with - it, and it doesn't appear to harm asterisk's operation, and may - prevent a crash. I'll store it in 1.2, as we have shut down - support on 1.2, but since I developed the patch before support - finished, and it might affect 1.4 and trunk, I'm going ahead with - it. ........ ................ - -2007-08-02 18:05 +0000 [r77940-77944] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 77943 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77943 | russell | 2007-08-02 13:04:43 -0500 (Thu, 02 Aug 2007) | - 9 lines Fix another race condition in the handling of dynamic - threads. If the dynamic thread timed out waiting for something to - do, but was acquired to perform an action immediately afterwords, - then wait on the condition again to give the other thread a - chance to finish setting up the data for what action this thread - should perform. Otherwise, if it immediately continues, it will - perform the wrong action. (reported on IRC by mihai, patch by me) - (related to issue #10289) ........ - - * channels/chan_iax2.c: Fix an issue that Simon pointed out to me - on IRC. There were cases in the trunk version of - find_idle_thread() where the old full frame processing - information was not cleared out. This would have caused full - frames to get deferred for processing by threads that weren't - actually processing frames for that call. Nice catch!! - - * /, channels/chan_iax2.c: Merged revisions 77939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77939 | russell | 2007-08-02 11:56:04 -0500 (Thu, 02 Aug 2007) | - 4 lines Add another sanity check to vnak_retransmit(). This check - ensures that frames that have already been marked for deletion - don't get retransmitted. (closes issue #10361, patch from mihai) - ........ - -2007-08-02 15:16 +0000 [r77891-77895] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 77894 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10358) ........ r77894 | qwell | 2007-08-02 10:15:45 -0500 - (Thu, 02 Aug 2007) | 5 lines Make sure that we show the correct - extension if dialed from a macro "From: 5555" rather than "From: - s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, - tested by sbisker ........ - - * /, channels/chan_skinny.c: Merged revisions 77890 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10291) ........ r77890 | qwell | 2007-08-01 17:28:56 -0500 - (Wed, 01 Aug 2007) | 4 lines Put in some additional debug - information for softkey/stimulus messages. Issue 10291, patch by - DEA. ........ - -2007-08-01 22:24 +0000 [r77889] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 77887 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77887 | russell | 2007-08-01 17:16:17 -0500 (Wed, 01 Aug 2007) | - 23 lines Fix some race conditions which have been causing weird - problems in chan_iax2. The most notable problem is that people - have been seeing storms of VNAK frames being sent due to really - old frames mysteriously being in the retransmission queue and - never getting removed. It was possible that a dynamic thread got - created, but did not acquire its lock before the thread that - created it signals it to perform an action. When this happens, - the thread will sleep until it hits a timeout, and then get - destroyed. So, the action never gets performed and in some cases, - means a frame doesn't get transmitted and never gets freed since - the scheduler never gets a chance to reschedule transmission. - Another less severe race condition is in the handling of a - timeout for a dynamic thread. It was possible for it to be - acquired to perform at action at the same time that it hit a - timeout. When this occurs, whatever action it was acquired for - would never get performed. (patch contributed by Mihai and - SteveK) (closes issue #10289) (closes issue #10248) (closes issue - #10232) (possibly related to issue #10359) ........ - -2007-08-01 22:19 +0000 [r77888] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 77886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77886 | tilghman | 2007-08-01 17:14:47 -0500 (Wed, 01 Aug 2007) - | 2 lines Voicemail with ODBC_STORAGE defined does not compile - cleanly (missing def) ........ - -2007-08-01 21:12 +0000 [r77879-77884] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 77883 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77883 | qwell | 2007-08-01 16:08:42 -0500 (Wed, 01 Aug - 2007) | 7 lines Fix an issue that caused one-way audio on some - newer devices (specifically the 7921), due to sending packets in - the wrong order during hangup. Also make sure we clear - tones/messages on the correct line/instance. Issue 10291, patch - by DEA, tested by sbisker and myself. ........ - - * apps/app_queue.c, doc/tex/queuelog.tex: Add the Ring time in the - CONNECT on the queue_log and on the Manager event AgentConnect - Closes issue #10349, patch by eliel - -2007-08-01 19:37 +0000 [r77864-77878] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, configure, configure.ac, main/asterisk.c: Instead of - adding the SOLARIS check to each HAVE_SYSINFO check let's just - make the sysinfo autoconf logic a bit pickier about what it - considers a usable sysinfo. - - * main/pbx.c, main/asterisk.c: Solaris does not have a sysinfo like - we know of on Linux. - - * configure, configure.ac: Don't look for /dev/urandom when cross - compiling. Just assume it is not available. - - * /, utils/smsq.c, channels/chan_iax2.c, - include/asterisk/threadstorage.h, channels/chan_mgcp.c, - apps/app_voicemail.c: Merged revisions 77869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77869 | file | 2007-08-01 14:56:59 -0300 (Wed, 01 Aug 2007) | 2 - lines Add some fixes for building on Solaris. ........ - - * /, main/utils.c: Merged revisions 77867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77867 | file | 2007-08-01 14:52:11 -0300 (Wed, 01 Aug 2007) | 2 - lines Whoops, I meant R_5 not R5. ........ - - * /, configure, configure.ac: Merged revisions 77865 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77865 | file | 2007-08-01 14:42:52 -0300 (Wed, 01 Aug - 2007) | 2 lines And for my last trick... make sure that if - gethostbyname_r is exported by a library that it is used. - ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: Merged revisions 77863 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77863 | file | 2007-08-01 14:22:35 -0300 (Wed, 01 Aug 2007) | 2 - lines Extend autoconf logic to determine which version of - gethostbyname_r is on the system. ........ - -2007-08-01 15:39 +0000 [r77858] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, main/autoservice.c, main/pbx.c, - apps/app_osplookup.c, channels/chan_local.c, - channels/chan_vpb.cc, apps/app_meetme.c, res/res_features.c, - apps/app_zapras.c, apps/app_macro.c, pbx/pbx_dundi.c, - apps/app_queue.c: Convert code that checks the _softhangup member - of ast_channel directory to use the ast_check_hangup() funciton. - This function takes scheduled hangups into account. (closes issue - #10230, patch by Juggie) - -2007-08-01 15:28 +0000 [r77857] Joshua Colp <jcolp@digium.com> - - * main/cli.c: Convert CLI helpers list to rwlist. - -2007-08-01 14:09 +0000 [r77853-77855] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 77854 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77854 | mmichelson | 2007-08-01 09:08:57 -0500 (Wed, 01 Aug - 2007) | 8 lines Fixes an issue I introduced to queues wherein a - queue with joinempty=yes would kick people out of the queue - because of erroneously thinking the 'n' option was in use. - (closes issue #10320, reported by jfitzgibbon, patched by me, - tested by blitzrage and me) Thank you blitzrage for all the - testing you've done lately with queues! It's much appreciated! - ........ - - * /, apps/app_queue.c: Merged revisions 77852 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77852 | mmichelson | 2007-08-01 08:59:59 -0500 (Wed, 01 Aug - 2007) | 7 lines If a queue uses dynamic realtime members, then - the member list should be updated after each attempt to call the - queue. This fixes an issue where if a caller calls into a queue - where no one is logged in, they would wait forever even if a - member logged in at some point. (closes issue #10346, reported by - and tested by blitzrage, patched by me) ........ - -2007-08-01 04:36 +0000 [r77851] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c: Twould help if we actually defined ->mod before - comparing against it (reported and fixed by Juggie via IRC). - -2007-07-31 21:33 +0000 [r77847] Steve Murphy <murf@digium.com> - - * /, contrib/scripts/ast_grab_core: Merged revisions 77844 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r77844 | murf | 2007-07-31 14:59:10 -0600 (Tue, - 31 Jul 2007) | 9 lines Merged revisions 77842 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 - line This probably isn't super-general, but it's a first stab at - using kill -11 to generate a core file instead of gcore. ........ - ................ - -2007-07-31 18:50 +0000 [r77834-77838] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_lock.c, CHANGES: Add some documentation detailing an - aspect of dialplan functions, as requested by Russell - - * funcs/func_lock.c (added), UPGRADE.txt: Add func_lock, which - creates dialplan mutexes, and note that the Macro apps are now - deprecated. (Closes issue #10264) - -2007-07-31 16:21 +0000 [r77833] Joshua Colp <jcolp@digium.com> - - * /, include/asterisk/speech.h, res/res_speech.c: Merged revisions - 77831 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77831 | file | 2007-07-31 13:17:09 -0300 (Tue, 31 Jul 2007) | 2 - lines Add a flag to the speech API that allows an engine to set - whether it received results or not. ........ - -2007-07-31 15:59 +0000 [r77829] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: thanks to Russel, for pointing out that the - dialoglist_lock/unlock routines also need to be macros if - DETECT_DEADLOCKS is set - -2007-07-31 15:54 +0000 [r77828] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/cflags.xml, /: Merged revisions 77827 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77827 | kpfleming | 2007-07-31 10:53:42 -0500 (Tue, 31 - Jul 2007) | 2 lines DETECT_DEADLOCKS can't be enabled without - DEBUG_THREADS or it does nothing ........ - -2007-07-31 15:22 +0000 [r77825] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 77824 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul - 2007) | 6 lines This patch makes Asterisk send 100 Trying - provisional responses upon receipt of re-invites. This makes it - so that if there are two or more Asterisk servers between - endpoints, the Asterisk servers will not keep retransmitting the - re-invites. (closes issue #10274, reported by cstadlmann, patched - by me with approval from file) ........ - -2007-07-31 15:01 +0000 [r77819-77821] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: there is no use in having functions that - have no code in them, and hide the locking info when - DEBUG_THREADS is enabled... i could have fixed this to be - dependent on DEBUG_THREADS, but it would be just as easy for - someone to add their test/debugging code to the macros as it - would have been to the functions - - * channels/chan_sip.c: use a different method for overriding the - send_digit_begin pointer, as the old one fails to compile on my - 64-bit system with gcc-4.1 and --enable-dev-mode turned on - - * apps/app_senddtmf.c: umm... let's build with --enable-dev-mode, - mmkay? - -2007-07-31 03:32 +0000 [r77810] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: Discovered in experiments on core files: if - you wrap the lock and unlock calls with sip_pvt_lock and - sip_pvt_unlock, you lose the tracing info you would normally get - via DETECT_DEADLOCKS; so I turn these two functions into macros - when DETECT_DEADLOCKS is called. This way, you get meaningful - stuff in the file and func slots in the lock_info struct. - -2007-07-31 01:10 +0000 [r77808] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c, apps/app_dictate.c, apps/app_record.c, - apps/app_authenticate.c, apps/app_sayunixtime.c, - apps/app_userevent.c, apps/app_chanisavail.c, apps/app_image.c, - apps/app_followme.c, apps/app_controlplayback.c, - funcs/func_enum.c, funcs/func_odbc.c, apps/app_minivm.c, - res/res_agi.c, apps/app_amd.c, apps/app_url.c, - apps/app_directory.c, apps/app_rpt.c, apps/app_parkandannounce.c, - apps/app_read.c, funcs/func_timeout.c, apps/app_page.c, - apps/app_festival.c, apps/app_privacy.c, - apps/app_waitforsilence.c, apps/app_disa.c, apps/app_transfer.c, - apps/app_talkdetect.c, apps/app_queue.c, apps/app_playback.c, - res/res_monitor.c, apps/app_speech_utils.c, funcs/func_curl.c, - funcs/func_channel.c, funcs/func_cdr.c, apps/app_sendtext.c, - apps/app_macro.c, apps/app_sms.c, apps/app_senddtmf.c, - apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_stack.c, - apps/app_voicemail.c: Mostly cleanup of documentation to - substitute the pipe with the comma, but a few other formatting - cleanups, too. - -2007-07-30 20:42 +0000 [r77801] Joshua Colp <jcolp@digium.com> - - * main/dial.c, include/asterisk/dial.h: Add support for call - forwarding and timeouts to the dialing API. - -2007-07-30 20:36 +0000 [r77797-77800] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Change another unnecessary use of the - increment operator to explicitly set the var to 1 - - * channels/chan_iax2.c: Explicitly set a variable to 1 instead of - using the increment operator. - - * /, channels/chan_iax2.c: Merged revisions 77794 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) | - 8 lines Fix an issue that could potentially cause corruption of - the global iax frame queue. In the network_thread() loop, it - traverses the list using the AST_LIST_TRAVERSE_SAFE macro. - However, to remove an element of the list within this loop, it - used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I - believe could leave some of the internal variables of the SAFE - macro invalid. Mihai says that he already made this change in his - local copy and it didn't help his VNAK storm issues, but I still - think it's wrong. :) ........ - -2007-07-30 20:19 +0000 [r77796] Jason Parker <jparker@digium.com> - - * /, main/say.c: Merged revisions 77795 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10083) ........ r77795 | qwell | 2007-07-30 15:17:08 -0500 - (Mon, 30 Jul 2007) | 6 lines Applications like SayAlpha() should - not hang up the channel if you request an "unknown" character - such as a comma. Instead, skip the character and move on. Issue - 10083, initial patch by jsmith, modified by me. ........ - -2007-07-30 19:42 +0000 [r77793] Luigi Rizzo <rizzo@icir.org> - - * main/channel.c: print formats as 0x%x instead of %d in a warning - message. Being bitmasks, it is a lot easier to read this way. - -2007-07-30 19:39 +0000 [r77789-77792] Russell Bryant <russell@digium.com> - - * res/res_agi.c: Fix the return value of ast_agi_fdprintf() to - include the result from ast_carefulwrite() - - * res/res_agi.c: Improve ast_agi_fdprintf() by using the ast_str() - API. * Use a thread local ast_str for building the string that - will be written out to the console for debug, and to the FD for - the AGI itself, instead of allocating a buffer on the heap every - time the function is called. * Use the information contained - within the ast_str to determine how many bytes need to be written - instead of calling strlen(). - - * main/manager.c: Remove an XXX comment noting that it would be - nice for a declaration to be inside of a function. (Yes, it - would!) Replace it with a note that explains why it can't be done - using the way that the AST_THREADSTORAGE macro is currently - defined. - - * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 77788 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | - 10 lines (closes issue #10279) Reported by: seanbright Patches: - res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright - (license 71) res_agi.carefulwrite.trunk.07252007.patch uploaded - by seanbright (license 71) Allow the "agi_network: yes" line to - be printed out in the AGI debug output. Also, allow partial - writes to be handled when writing out this line just like it is - for all of the others. ........ - -2007-07-30 19:11 +0000 [r77787] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/agi.h, res/res_agi.c: Cleanup of res_agi, - ensuring thread safety (closes issue #10288) - -2007-07-30 18:56 +0000 [r77786] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 77785 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | - 3 lines file and I both committed changes for issue #10301. - Remove a duplicated assignment to restore the original value of - the previous channel. ........ - -2007-07-30 18:45 +0000 [r77784] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 77783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r77783 | tilghman | 2007-07-30 13:43:55 -0500 - (Mon, 30 Jul 2007) | 10 lines Merged revisions 77782 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 - Jul 2007) | 2 lines Revert change in revision 71656, even though - it fixed a bug, because many people were depending upon the - (broken) behavior. ........ ................ - -2007-07-30 17:31 +0000 [r77781] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 77780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | - 16 lines (closes issue #10301) Reported by: fnordian Patches: - asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) - Additional changes by me Fix some problems in - channel_find_locked() which can cause an infinite loop. The - reference to the previous channel is set to NULL in some cases. - These changes ensure that the reference to the previous channel - gets restored before needing it again. I'm not convinced that the - code that is setting it to NULL is really the right thing to do. - However, I am making these changes to fix the obvious problem and - just leaving an XXX comment that it needs a better explanation - that what is there now. ........ - -2007-07-30 17:12 +0000 [r77772-77779] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 77778 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 - lines (closes issue #10327) Reported by: kkiely Instead of - directly mucking with the extension/context/priority of the - channel we are transferring when it has a PBX simply call - ast_async_goto on it. This will ensure that the channel gets - handled properly and sent to the right place. ........ - - * apps/app_followme.c: Minor clean up of app_followme. - - * main/channel.c, /: Merged revisions 77771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 - lines (closes issue #10301) Reported by: fnordian Patches: - asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) - Restore previous behavior where if we failed to lock the channel - we wanted we would return to exactly the same point as if we had - just reentered the function. ........ - -2007-07-30 15:22 +0000 [r77770] Russell Bryant <russell@digium.com> - - * cdr/cdr_adaptive_odbc.c: Resolve some compiler warnings so that I - can build under dev mode - -2007-07-30 14:53 +0000 [r77769] Joshua Colp <jcolp@digium.com> - - * /, apps/app_macro.c: Merged revisions 77768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r77768 | file | 2007-07-30 11:51:44 -0300 (Mon, - 30 Jul 2007) | 12 lines Merged revisions 77767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 - lines (closes issue #10334) Reported by: ramonpeek Pass through - the return value from macro_exec through the MacroIf application. - ........ ................ - -2007-07-30 10:55 +0000 [r77616-77766] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: minor code rearrangements: + place the link - field at the beginning of struct sip_pvt, and not somewhere in - the middle; + in __sip_reliable_xmit, remove a duplicate - assignment, and put the statements in a more logical order (i.e. - first copy the payload and associated info, then copy arguments - from the caller, then finish initializing the headers...) nothing - to backport. - - * channels/chan_sip.c: rename handle_request() to - handle_incoming(), as the former was misleading - the function - deals with all incoming packets, be them requests or responses. - - * channels/chan_sip.c: move some dialog-only flags to proper - variables, namely SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, - SIP_PAGE2_NOTEXT, SIP_PAGE2_OUTGOING_CALL These are seldom used - so the diff is relatively small. Note that 'OUTGOING_CALL' is - dangerously similar to another dialog flag, 'SIP_OUTGOING', so - the description will need to clarify the different meaning of the - two. Also note that the description of NOTEXT is a bit unclear - - does it mean we don't support it, or 'not requested or not - supported' ? On passing fix a comment referring to video instead - of text. Finally, mark with XXX a possibly misleading debugging - message. (maybe the latter is worth backporting). - - * channels/chan_sip.c: use a function, cli_yesno(), to produce the - output Yes or No for CLI lines. This helps maintaining - consistency on output, slightly improves readability, and maybe - one day will make it easier to translate the output in other - languages (though i have a hard time believing that a CLI user - who needs 'yes' and 'no' to be translated can actually figure out - what he/she is doing!) - - * channels/chan_sip.c: move the two remaining peer flags to proper - variables. - - * channels/chan_sip.c: move RT_FROMCONTACT to a proper sip_peer - field. - - * channels/chan_sip.c: Move some global 'flags' to individual - variables. Start putting these variables in a single struct - (called 'sip_cfg' for the time being, but it could as well be - 'global' or some other name) so it is easy, when reading the - code, to figure out what they are for. The downside of using - struct fields instead of individual global variables is that the - compiler cannot tell if there are unused fields. But the - advantage of not cluttering the namespace and manilpulating all - these variables at once certainly overcome the disadvantagess. - Nothing to backport, again. - - * channels/chan_sip.c: minor simplification of a conditional - statement - - * channels/chan_sip.c: build the version of sip_tech with no - send_digit_begin at load time instead of duplicating the - initializer. This should remove the risk of forgetting fields in - the initializer. - - * channels/chan_sip.c: remove bit position from description of - SIP_* flags. use AST_FORMAT_AUDIO_MASK instead of playing with - AST_FORMAT_MAX_AUDIO to determine audio formats. There is a - dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call() which - surely needs fixing, namely: /* mask request with some set of - allowed formats. * XXX this needs to be fixed. * The original - code uses AST_FORMAT_AUDIO_MASK, but it is * unclear what to use - here. We have global_capabilities, which is * configured from - sip.conf, and sip_tech.capabilities, which is * hardwired to all - audio formats. */ The latter is possibly something to backport - when fixed. - - * channels/chan_sip.c: back on cleaning up the usage of flags. Move - together flags used in the same way (e.g. dialog only, - dialog-peer, ...) so it will become easier to deal with them in a - more systematic way. This is being done in stages so it will be - easier to detect breakage, if any should occur. - - * channels/chan_sip.c: more documentation on internal - representation of incoming SIP messages. Remove definitions for - now-unused flags, and add references to print routines for other - flags. - - * channels/chan_sip.c: make register_unref() return NULL so it is - easy to cleanup the original pointer while calling the function. - on passing add some comments on one of the places where it is - used, and explain why it is safe there. again, a no-op for - practical purposes. - - * channels/chan_sip.c: add some documentation to auto_congest(), - and some dialog_ref/unref (they are a no-op at the moment). Also - clean a pointer after freeing memory to avoid dangling - references, and write a for() loop in canonical form. In - practice, everything in this commit is a no-op. - - * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls - - * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls - - * channels/chan_sip.c: start introducing hooks for reference counts - on dialog descriptors. This commit is, for all practical - purposes, a no-op, as it only introduces the dialog_ref() and - dialog_unref() methods, and uses them in a few places (not all - the places where they would be needed). The goal is to start - annotating the code with these calls, so the transition to a - proper container will be easier. Nothing to backport. - - * channels/chan_sip.c: remove an unused string - - * channels/chan_sip.c: simplify a conditional expression using S_OR - - * channels/chan_sip.c: make use of received= and rport= fields in - sip replies. In a nutshell, these fields are used to tell a sip - entity the address and port its request came from, and are - extremely useful in the presence of NATs, especially with - symmetric NATs where STUN is totally ineffective. This patch - stores the address and port in the 'ourip' field of the dialog - descriptor, so they can be reused in subsequent transactions. As - it is, it works well for things like REGISTER requiring - authentication, because the second REGISTER request (with auth - credentials) will carry the correct address. Maybe it can also be - useful, in case of an address change, to do one or both of the - following: + propagate the new address to the parent user/peer - descriptor so that new dialogs will use the correct address from - the beginning. This is trivial to implement, I am just waiting - for feedback on this. + re-issue a request in case of an address - change. This a lot less trivial, maybe unnecessary, and probably - covered by the previous item. I would seriously consider this - patch for addition to 1.4 and 1.2. The code is very little - intrusive, and it would solve in a correct way the nat traversal - problems for which externip/externaddr/stunaddr are only a - partial and expensive workaround. - -2007-07-27 23:21 +0000 [r77572-77603] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Some ODBC drivers don't set the - CHAR_OCTET_LENGTH field correctly. - - * Makefile: Target asterisk.pdf stopped building when the build was - moved to the doc directory. - - * /, res/res_odbc.c: Merged revisions 77571 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77571 | tilghman | 2007-07-27 13:15:58 -0500 (Fri, 27 Jul 2007) - | 2 lines Missing newline ........ - -2007-07-27 17:05 +0000 [r77537-77541] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 77540 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77540 | file | 2007-07-27 14:04:08 -0300 (Fri, 27 Jul 2007) | 6 - lines (closes issue #10310) Reported by: prashant_jois Patches: - cdr_pgsql.patch uploaded by prashant (license 114) Finish the - Postgresql connection after the log messages are printed so we - don't access invalid memory. ........ - - * channels/chan_sip.c: Turn 4 lines of code into 1 line that does - the same thing. - - * /, channels/chan_sip.c: Merged revisions 77536 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6 - lines (closes issue #10323) Reported by: julianjm Patches: - chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm - (license 99) Clear ONHOLD flag when decrementing the onHold peer - count. If we did not do this the count may keep decreasing. - ........ - -2007-07-27 16:20 +0000 [r77534] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_config.c: 'dialplan save' shouldn't be converting '|' - back to ',' anymore. - -2007-07-27 15:46 +0000 [r77520] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, pbx/pbx_ael.c: These fixes take care of two - problems: a complaint in asterisk-dev that goto's aren't working - in trunk, a side effect of the move to commas as arg seps in apps - and funcs; and a problem I spotted myself with dial's 'e' option, - where gotos were off by one, because I forgot to set the AUTOLOOP - flag in the peer channel. - -2007-07-27 14:31 +0000 [r77491] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 77490 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul - 2007) | 3 lines "re-invite" was misspelled ........ - -2007-07-26 23:20 +0000 [r77461] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 77460 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 - lines (closes issue #10302) Reported by: litnialex If a DTMF end - frame comes from a channel without a begin and it is going to a - technology that only accepts end frames (aka INFO) then use the - minimum DTMF duration if one is not in the frame already. - ........ - -2007-07-26 22:17 +0000 [r77432] Kevin P. Fleming <kpfleming@digium.com> - - * /, doc/tex/mp3.tex, sounds/Makefile: Merged revisions 77424,77429 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007) - | 2 lines use new canonical name for download server ........ - r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007) - | 2 lines change protocol for downloads as well ........ - -2007-07-26 21:24 +0000 [r77411] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 77410 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) | - 10 lines AST_DEVMODE was defined in trunk, but not in 1.4. When - Asterisk is compiled under dev mode, AST_DEVMODE will get defined - in buildopts.h. Change 1.4 to define it in the same way that - trunk does. Also, revert the change that added this define in the - Makefile The advantage to doing it this way is that buildopts.h - gets installed when you install Asterisk. Then, when building any - out of tree modules, or building asterisk-addons, these modules - know which options the rest of Asterisk was built with. ........ - -2007-07-26 20:39 +0000 [r77381] Mark Michelson <mmichelson@digium.com> - - * Makefile, /, main/logger.c: Merged revisions 77380 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 - Jul 2007) | 7 lines Fixes to get ast_backtrace working properly. - The AST_DEVMODE macro was never defined so the majority of - ast_backtrace never attempted compilation. The makefile now - defines AST_DEVMODE if configure was run with --enable-dev-mode. - Also, changes were made to acccomodate 64 bit systems in - ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for - their roles in allowing me to get this committed ........ - -2007-07-26 19:33 +0000 [r77349-77351] Tilghman Lesher <tlesher@digium.com> - - * /, main/logger.c: Merged revisions 77350 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007) - | 2 lines Missed one ........ - - * /, main/logger.c: Merged revisions 77348 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007) - | 2 lines Oops, that builtin define should be all-lowercase. - ........ - -2007-07-26 18:31 +0000 [r77319] Mark Michelson <mmichelson@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 77318 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul - 2007) | 8 lines Two consecutive calls to PQfinish could occur, - meaning free gets called on the same variable twice. This patch - sets the connection to NULL after calls to PQfinish so that the - problem does not occur. Also in this patch, prashant_jois - informed me that it is safe to pass a null pointer to PQfinish, - so I have removed the check for conn's existence from - my_unload_module. (closes issue 10295, reported by junky, patched - by me with input from prashant_jois) ........ - -2007-07-26 15:49 +0000 [r77268-77299] Russell Bryant <russell@digium.com> - - * main/udptl.c, res/res_features.c, main/say.c, - codecs/codec_adpcm.c, apps/app_alarmreceiver.c, - cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, - main/indications.c, main/config.c, main/loader.c, res/res_smdi.c, - pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_zapscan.c, - apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c, - apps/app_amd.c, cdr/cdr_odbc.c, res/res_speech.c, - apps/app_dial.c, codecs/codec_g722.c, funcs/func_timeout.c, - codecs/codec_speex.c, channels/chan_agent.c, codecs/codec_g726.c, - channels/iax2-provision.c, apps/app_db.c, channels/chan_misdn.c, - main/srv.c, apps/app_waitforring.c, apps/app_macro.c, - apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_vpb.cc, - apps/app_meetme.c, res/res_snmp.c, codecs/codec_gsm.c, - res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c, - res/res_jabber.c, main/channel.c, main/cdr.c, - channels/chan_phone.c, main/dial.c, res/res_config_odbc.c, - main/manager.c, funcs/func_odbc.c, res/res_agi.c, main/app.c, - main/image.c, apps/app_rpt.c, apps/app_parkandannounce.c, - channels/chan_mgcp.c, apps/app_adsiprog.c, apps/app_while.c, - codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c, - channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c, - main/translate.c, codecs/codec_alaw.c, apps/app_waitforsilence.c, - res/res_crypto.c, apps/app_queue.c, apps/app_getcpeid.c, - channels/chan_oss.c, main/rtp.c, apps/app_flash.c, - main/abstract_jb.c, main/file.c, channels/chan_h323.c, - codecs/codec_ulaw.c, pbx/pbx_dundi.c, apps/app_sms.c, - pbx/pbx_gtkconsole.c: Do a massive conversion for using the - ast_verb() macro (closes issue #10277, patches by mvanbaak) - Basically, this changes ... if (option_verbose > 2) - ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, - "Something\n"); - - * doc/tex/odbcstorage.tex, doc/tex/hardware.tex, doc/tex/mp3.tex, - doc/tex/channelvariables.tex, doc/tex/qos.tex, - doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex, - doc/tex/dundi.tex, doc/tex/enum.tex, doc/tex/asterisk-conf.tex, - doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/imapstorage.tex, - doc/tex/privacy.tex, LICENSE, doc/tex/app-sms.tex, - doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Merge a big batch of - documentation fixes for escaping, marking URLs, places where - verbatim text went off the end of the page on the PDF, and - various other improvements (closes issue #10307, IgorG) - - * channels/chan_sip.c: Revert some changes to call abs() on the - result of ast_random(). * random() is defined to return a - positive result, and now ast_random() will always do so as well - - * main/utils.c: Ensure that the read from /dev/urandom returns a - positive result (closes issue #10308, reported by yehavi, patched - by me) - -2007-07-26 13:19 +0000 [r77267] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: Things expecting a positive result from - ast_random() should not be surprised (closes #10308) - -2007-07-26 13:10 +0000 [r77266] Russell Bryant <russell@digium.com> - - * main/rtp.c: Add a link to the list of assigned RTP payload types - for convenience. - -2007-07-26 05:35 +0000 [r77233-77248] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: document how the RTP marker bit is passed for video - frames, and why this does not overwrite useful information. - - * main/rtp.c: add an entry for h263plus in an empty slot of the rtp - types. - -2007-07-26 01:33 +0000 [r77217-77218] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: The upgrade of application argument separators - to comma has an effect on AEL; I commented out the code that - substitutes commas with vertbars, so we can get apps to parse - their args correctly. - - * apps/app_meetme.c: Merged revisions 77191 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 - line This fix solves problem with intense squelch noise when - someone joins conf in bug 9430; We repro'd the problem with - meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm - applying it. It looks like playing the recorded username will - louse up the next thing played into the channel. Josh rearranged - the code so as to start things over before playing data directly - into the conference. ........ - -2007-07-25 22:18 +0000 [r77182] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 77176 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul - 2007) | 4 lines (closes issue #10303) Reported by: jtodd Add - SPEECH_DTMF_TERMINATOR variable so the user can specify the digit - to terminate a DTMF string with. If none is specified then no - terminator will be used. ........ - -2007-07-25 21:58 +0000 [r77156] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_iax2.c: silence a warning in ast-devmode on a - potentially uninitialized var. At first sight (but the function - is very large so i am not 100% sure) the code seems correct, so - maybe my compiler is just not smart enough to figure that out at - the optimization level it has. Not worthwhile merging to 1.4 i - believe. - -2007-07-25 21:53 +0000 [r77155] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: Merged revisions 77154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul - 2007) | 3 lines chan->emulate_dtmf_duration is an unsigned int, - not a signed int, so use %u instead of %d in the format string - ........ - -2007-07-25 17:16 +0000 [r77072] Joshua Colp <jcolp@digium.com> - - * /, configure, acinclude.m4: Merged revisions 77071 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul - 2007) | 2 lines Fix autoconf logic for finding OpenH323 when it - is not in the first place searched (/usr/share/openh323). - ........ - -2007-07-25 14:13 +0000 [r77023-77054] Luigi Rizzo <rizzo@icir.org> - - * main/translate.c: change the debug level to 3 for an exceedingly - annoying message (3-deep nested loop) - - * main/rtp.c: Merged revisions 77022 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 - lines set the sequence number in a frame for all frame types - ........ - -2007-07-25 01:06 +0000 [r76985] Russell Bryant <russell@digium.com> - - * CHANGES: remove a couple of entries that got duplicated and snuck - into the SIP section. Also, align the NAT/STUN entry with the - others. - -2007-07-25 00:34 +0000 [r76984] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 76983 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue, - 24 Jul 2007) | 9 lines Merged revisions 76978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 - line this fixes bug 10293, where the error message because - defaultzone or loadzone was not defined was confusing ........ - ................ - -2007-07-24 22:13 +0000 [r76874-76940] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 76937 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r76937 | tilghman | 2007-07-24 17:12:43 -0500 - (Tue, 24 Jul 2007) | 10 lines Merged revisions 76934 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 - Jul 2007) | 2 lines Oops, res contains the error code, not errno. - I was wondering why a mutex was reporting "No such file or - directory"... ........ ................ - - * build_tools/cflags.xml: Add the flag to trigger an intentional - crash on mutex errors - - * doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/jitterbuffer.tex, - doc/tex/odbcstorage.tex, doc/tex/hardware.tex, - doc/tex/privacy.tex, doc/tex/billing.tex, doc/tex/ael.tex, - doc/tex/channelvariables.tex, doc/tex/qos.tex, - doc/tex/realtime.tex, doc/tex/asterisk.tex, doc/tex/queuelog.tex: - Fix escaping and some of the formattting (closes issue #10285) - -2007-07-24 17:43 +0000 [r76841-76852] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Revert trivial whitespace change (for - testing) - - * channels/chan_skinny.c: Trivial whitespace change to test - comitting... - -2007-07-24 17:05 +0000 [r76807] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 76803 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 - lines Don't create the Asterisk channel until we are starting the - PBX on it. (ASA-2007-018) ........ - -2007-07-24 16:42 +0000 [r76804] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 76801 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul - 2007) | 13 lines Added a membercount variable to call_queue - struct which keeps track of the number of logged in members in a - particular queue. This makes it so that the 'n' option for - Queue() can act properly depending on which strategy is used. If - the strategy is roundrobin, rrmemory, or ringall, we want to ring - each phone once before moving on in the dialplan. However, if any - other strategy is used, we will only ring one phone since it - cannot be guaranteed that a different phone will ring on - subsequent attempts to ring a phone. As a side effect of this, - the QUEUE_MEMBER_COUNT dialplan function now just reads the - membercount variable instead of traversing through the member - list to figure out how many members there are. Special thanks to - blitzrage for helping to test this out. (closes issue #10127, - reported by bcnit, patched by me, tested by blitzrage) ........ - -2007-07-24 16:09 +0000 [r76791] Joshua Colp <jcolp@digium.com> - - * sounds/Makefile: Don't download/install the sound packages if - already installed. - -2007-07-24 15:35 +0000 [r76785] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: The chan_skinny Dial() syntax was funky. - You had to do Dial(Skinny/line@device) This allows you to just - Dial(Skinny/line), as long as line isn't ambiguous. Note that - this does not remove or deprecate the "old" syntax, as it's still - quite useful - even moreso if shared lines get implemented. - Initial patch by me, with some changes and suggestions from - wedhorn. (closes issue #10263) - -2007-07-24 14:49 +0000 [r76755-76770] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: two small fixes when using stun (reported by - Marta Carbone): + externexpire was not initialized properly; + - stunaddr was not handled properly on a sip reload - - * CHANGES: add documentation on nat/stun support in chan_sip - -2007-07-24 02:59 +0000 [r76710-76712] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Move manager users list over to an rwlist. - - * res/res_agi.c: You need to put static in front of a static RWLIST - declaration to make it really static... and don't call - AST_RWLIST_HEAD_DESTROY on a statically declared list. - - * main/manager.c: Don't bother calling AST_RWLIST_EMPTY on a list - before AST_RWLIST_TRAVERSE, it's just a double check. - -2007-07-23 22:41 +0000 [r76707-76709] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 76708 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) - | 4 lines It was our stated intention for 1.4 that files created - in app_voicemail should depend upon the umask. Unfortunately, - mkstemp() creates files with mode 0600, regardless of the umask. - This corrects that deficiency. ........ - - * include/asterisk/agi.h, res/res_agi.c: Enhance AGI with several - fixes: - Makes the structures handling external AGI commands a - bit more thread-safe - Makes AGI transparently work with both - live and hungup channels - DeadAGI is hence no longer necessary - and is deprecated - CLI bug fixes - Commands will refuse to run - if the channel is dead and the command is nonsensical for dead - channels. - -2007-07-23 21:42 +0000 [r76706] Joshua Colp <jcolp@digium.com> - - * res/res_crypto.c: Clean up res_crypto module. It now uses an - rwlist to keep the keys and it should also be thread safe now. - -2007-07-23 20:27 +0000 [r76703-76704] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c, UPGRADE.txt: Missed one conversion to comma - delimiter (thanks, Juggie) and add documentation on the change to - the Local channel name. - - * funcs/func_rand.c, apps/app_readfile.c, channels/chan_local.c, - apps/app_record.c, funcs/func_env.c, funcs/func_strings.c, - funcs/func_vmcount.c, include/asterisk/aes.h, funcs/func_logic.c, - apps/app_exec.c, apps/app_controlplayback.c, funcs/func_odbc.c, - apps/app_skel.c, apps/app_zapras.c, apps/app_url.c, - apps/app_externalivr.c, apps/app_parkandannounce.c, - apps/app_dial.c, main/pbx.c, apps/app_page.c, - apps/app_softhangup.c, UPGRADE.txt, funcs/func_cut.c, - apps/app_talkdetect.c, apps/app_queue.c, funcs/func_realtime.c, - include/asterisk/app.h, apps/app_channelredirect.c, - apps/app_macro.c, pbx/pbx_config.c, apps/app_verbose.c, - apps/app_chanspy.c, funcs/func_callerid.c, apps/app_voicemail.c: - Merge the dialplan_aesthetics branch. Most of this patch simply - converts applications using old methods of parsing arguments to - using the standard macros. However, the big change is that the - really old way of specifying application and arguments separated - by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the - way that has been recommended since long before 1.0 will become - the only method available (e.g. NoOp(foo,bar). - -2007-07-23 19:00 +0000 [r76657] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 76656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul - 2007) | 3 lines Fix some incorrect softkey labels in messages. - Don't try to play dialtone in some unimplemented features. - ........ - -2007-07-23 18:31 +0000 [r76655] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 76654 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, - 23 Jul 2007) | 12 lines Merged revisions 76653 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 - lines (closes issue #5866) Reported by: tyler Do not force - channel format changes when a generator is present. The generator - may have changed the formats itself and changing them back would - cause issues. ........ ................ - -2007-07-23 17:58 +0000 [r76621] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 76620 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10276) ........ r76620 | qwell | 2007-07-23 12:57:53 -0500 - (Mon, 23 Jul 2007) | 4 lines Don't try to queue up hold/unhold - frames on a non-existent channel. Issue 10276. ........ - -2007-07-23 17:49 +0000 [r76619] Joshua Colp <jcolp@digium.com> - - * /, apps/app_morsecode.c: Merged revisions 76618 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2 - lines Allow app_morsecode to build on PPC Linux by putting the - value of the digit char in an int. ........ - -2007-07-23 14:45 +0000 [r76564] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: add two missing entries in the replica of - the sip_tech that does not use DTMF BEGIN frames. 1.4 seems - correct (it does not have the two fields). However, as this bug - shows, the current way of creating the sip_tech replica is too - error-prone, one can easily forget to update one of the two - entries. Perhaps it would be better to create sip_tech_info - expliclty at module load, by doing sip_tech_info = sip_tech; - sip_tech_info.send_digit_begin = NULL (in this case, this is - something applicable to 1.4 as well). - -2007-07-23 14:38 +0000 [r76563] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, - 23 Jul 2007) | 14 lines Merged revisions 76560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 - lines (closes issue #10236) Reported by: homesick Patches: - rpid_1.4_75840.patch uploaded by homesick (license 91) Accept - Remote Party ID on guest calls. ........ ................ - -2007-07-23 14:37 +0000 [r76555-76562] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Mark str2dtmfmode() as currently unused to - resolve a compiler warning and allow building under dev mode - - * include/asterisk.h, res/res_snmp.c, channels/chan_sip.c, - res/res_crypto.c, res/res_convert.c, main/devicestate.c, - include/jitterbuf.h, res/res_config_sqlite.c, main/enum.c, - res/res_monitor.c, include/asterisk/file.h, - include/asterisk/doxyref.h, res/res_config_odbc.c, - res/res_indications.c, main/asterisk.c, res/res_clioriginate.c: - (closes issue #10271) Reported by: snuffy Patches: - doxygen-updates.diff uploaded by snuffy (license 35) Another big - batch of doxygen documentation updates - - * CHANGES: note the debug and verbose changes in CHANGES - - * include/asterisk/logger.h, main/pbx.c, main/logger.c, - include/asterisk/options.h, main/asterisk.c, main/cli.c: (closes - issue #10192) Reported by: bbryant Patches: - 20070720__core_debug_by_file.patch uploaded by bbryant (license - 36) (with some modifications by me) Tested by: russell, bbryant - This set of changes introduces the ability to set the core debug - or verbose levels on a per-file basis. Interestingly enough, in - 1.4, you have the ability to set core debug for a single file, - but that functionality was accidentally lost in the conversion of - the CLI commands to the new format. This patch improves upon what - was in 1.4 by letting you set it for more than 1 file, and by - also supporting verbose. *** Janitor Project *** This patch also - introduces a new macro, ast_verb(), which is similar to - ast_debug(). Setting the per file verbose value only works for - messages that use this macro. Converting existing uses of - ast_verbose() can be done like: if (option_debug > 2) - ast_verbose(VERBOSE_PREFIX_3 "Something useful\n"); ... - ast_verb(3, "Something useful\n"); - -2007-07-23 14:18 +0000 [r76547] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: introduce two functions, map_x_s() and - map_s_x(), to map between integers and strings using a single - translation table, and use them in a few places instead of ad-hoc - routines that duplicate the table. On passing, note that - REFER_CONFIRMED is never used, and add a few comments. Nothing to - backport here. - -2007-07-23 14:02 +0000 [r76524] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove an unused function to resolve a - compiler warning - -2007-07-23 13:46 +0000 [r76523] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Use autoconf - logic to determine byte swapping macro presence. This should now - also use other macros if present. - -2007-07-23 13:29 +0000 [r76521] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: move "sip prunte realtime ..." and "sip set - debug ... " to NEW_CLI style. - -2007-07-23 13:24 +0000 [r76520] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 76519 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul - 2007) | 6 lines (closes issue #10268) Reported by: mvanbaak - Patches: chan_skinny_openbsd.diff uploaded by mvanbaak (license - 7) Add another OS that has to use the Macros for byte ordering. - ........ - -2007-07-23 12:29 +0000 [r76486] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 76485 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76485 | russell | 2007-07-23 07:25:01 -0500 (Mon, 23 Jul 2007) | - 6 lines Use a signed integer for storing the number of bytes in - the packet read from the network. Using an unsigned value here - made it impossible to handle an error returned from recvfrom(). - Furthermore, in the case that recvfrom() did return an error, - this would cause a crash due to a heap overflow. (closes issue - #10265, reported by and fix suggested by timrobbins) ........ - -2007-07-23 03:10 +0000 [r76313-76467] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: Add some documentation on the sipregistry - states and the handling of the sip_register structures. This - commit only changes comments and whitespace. - - * channels/chan_sip.c: add a bit of comments on internal functions. - - * channels/chan_sip.c: rewrite "sip show {channels|subscriptions}" - CLI handler using the new-style cli format. No functional - changes, nothing to backport. - - * channels/chan_sip.c: Make sip_destroy() return NULL so the caller - can do things like foo = sip_destroy(foo); and reduce the chance - of bugs due to dangling pointers. Also remove a duplicate - prototype for the function. nothing to backport. - - * channels/chan_sip.c: add two comment blocks, one on reusing - nonces, and one on the handling of an 'authpeer' local variable. - - * channels/chan_sip.c: comment and slightly restructure - handle_request() in the part that handles responses, so that - there is a common exit point. Mark two places where probably we - could return -1 instead of 0 to report an error to the caller. - (change triggered by investigations on how the 'SIP_PKT_IGNORE' - field was used). nothing to backport from this commit - - * channels/chan_sip.c: remove unused argument from - handle_invite_replaces(), and also leftover SIP_PKT_* stuff from - the previous commit. - - * channels/chan_sip.c: Cleanup of flags used in struct sip_request, - moving them to individual variables. Apart from SIP_PKT_IGNORE - which was used a zillion times, the other two are used seldom. On - passing: - move the arrays to the end of struct sip_request, so a - (small) buffer overflow is less likely to overwrite the other - fields; - note that the 'ignore' argument to - handle_invite_replaces() is not used and should be removed (will - be done in a separate commit). Nothing to backport in this - change. - - * channels/chan_sip.c: move two per-packet flags to proper - variables. - - * channels/chan_sip.c: minor clarification on the usage of SIP_* - flags. Also correct some items that were misclassified. - - * channels/chan_sip.c: document the way sipdebug works, and - implement it through variables and not flags. NOTE: The old - behaviour (preserved in this commit) is that if sipdebug is set - in the config file, it can only be disabled by reloading the - config. I am not sure if this is accidental or voluntary, but it - is really unconvenient and I think it should be handled in the - same way as other options i.e. consider requests from the config - file or the cli (or the command line) to be fully equivalent and - act on the same status variable. - - * channels/chan_sip.c: move the SIP_REALTIME flag to a field in the - user/peer structure. - - * channels/chan_sip.c: Add a note to document how the temporary - 'pvt' should be initialized before using it. I am unclear on the - details right now so i hope someone can comment more. The obvious - (and lazy) approach would be to bzero() all of it (except for the - string pool), but isn't that too much work ? Feedback wanted - here... - -2007-07-21 14:39 +0000 [r76296] Joshua Colp <jcolp@digium.com> - - * include/asterisk/utils.h, configure, - include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Add - support for using /dev/urandom to get random numbers on systems - that support it. - -2007-07-21 09:35 +0000 [r76229-76279] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: whoops... was setting needdestroy on the - wrong dialog. (spotted by a diff with my own branch) - - * channels/chan_sip.c: more two more flags to proper variables: - ALREADYGONE and NEEDDESTROY. - - * channels/chan_sip.c: use explicit variables for things that don't - need to be stored in ast_flags. First victim is 'SIP_NO_HISTORY' - replaced by a 'do_history' field in the sip_pvt structure. - - * channels/chan_sip.c: Use ast_str_append() instead of - ast_build_string() to construct the sdp messages. Overall the - code is slightly more readable (because the string is fully - described by a single pointer), and more efficient (because the - length is stored explicitly so you don't need to do strlen()). (I - have been using this code for almost a year now.) I wish we had - infix string operators to do this sort of things! Nothing to - backport from this change. - -2007-07-21 01:25 +0000 [r76224] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: We have two 'technology' descriptors for a - SIP channel, so define and use a macro to determine whether we - are pointing to one of them, so when one goes away (or a new one - appears) we don't have to touch all the code. - -2007-07-21 01:08 +0000 [r76222] Steve Murphy <murf@digium.com> - - * apps/app_queue.c: One small documentation update made to - accompany 10154, the upgrading of the queue ringing to allow - periodic announcments - -2007-07-21 01:01 +0000 [r76221] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c, configs/sip.conf.sample: Enhance NAT support - as discussed on the -dev list, i.e.: + extensive documentation - changes both in sip.conf.sample and in the source; + allow - "externip" and "externhost" to include a port number as well; + - allow "bindaddr" to have a port number (making bindport - unnecessary, even though it is still present for backward - compatibility); + introduce the new "stunaddr" parameter to - specify an STUN server to be used from the main SIP socket; + - extend the "sip show settings" output to show all the above. - Internally: + change related data structures from struct in_addr - to struct sockaddr_in to store the port numbers as well; + - reorganize ast_sip_ouraddrfor() (should also be renamed to - sip_ouraddrfor() because it is not a generic API, though it might - become so if called with a socket as an additional argument, in - which case it can be moved elsewhere). As mentioned in the - documentation, media sessions still do not use STUN so the port - numbers may still be incorrect when Asterisk is behind a NAT On - passing, some of the debugging messages printing media addresses - are probably using the wrong values, but this will be - checked/fixed in a subsequent commit if needed. Part of the - following chunk in the function that handles a "sip reload" is - probably needed on previous versions as well, to avoid leaking - the memory used for the "localaddr" list: @@ -17244,13 +17274,17 - @@ /* Reset IP addresses */ memset(&bindaddr, 0, - sizeof(bindaddr)); + memset(&stunaddr, 0, sizeof(stunaddr)); + - memset(&internip, 0, sizeof(internip)); + /* Free memory for - local network address mask */ + ---> ast_free_ha(localaddr); - <----- memset(&localaddr, 0, sizeof(localaddr)); - memset(&externip, 0, sizeof(externip)); memset(&default_prefs, 0 - , sizeof(default_prefs)); - -2007-07-21 00:57 +0000 [r76220] Steve Murphy <murf@digium.com> - - * apps/app_queue.c: This update was supplied in 10154; to allow - announcemnts if the 'r' option (ringing) is provided. - -2007-07-20 22:25 +0000 [r76216] Jason Parker <jparker@digium.com> - - * configs/say.conf.sample, apps/app_playback.c: Add support for - default "say mode" (whether to use the "old" method or "new" - method. "new" method being config file) Add support for - autocomplete of "say load" CLI command. Patch by IgorG (closes - issue #10243) - -2007-07-20 21:41 +0000 [r76213] Steve Murphy <murf@digium.com> - - * /, sounds/Makefile: Merged revisions 76211 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76211 | murf | 2007-07-20 15:36:05 -0600 (Fri, 20 Jul 2007) | 1 - line This patch from 10249 is worth applying! It prevents - downloading sound files if they are already downloaded. Darn - Practical, if you ask me ........ - -2007-07-20 21:04 +0000 [r76175-76179] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 76174 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r76174 | qwell | 2007-07-20 15:32:55 -0500 (Fri, 20 Jul - 2007) | 2 lines It's possible for sub->owner to be NULL here if - you cancel the call immediately after/during sending a digit. - ........ - -2007-07-20 18:44 +0000 [r76140] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_directory.c: Merged revisions 76139 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul - 2007) | 6 lines When using users.conf for the entries in the - directory, if multiple users had the same last name, only the - first user listed would be available in the directory. (closes - issue #10200, reported by mrskippy, patched by me) ........ - -2007-07-20 18:28 +0000 [r76138] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 76132 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | - 6 lines Use the define that specifies the default length of an - artificially created DTMF digit in the ast_senddigit() function. - The define is set to 100ms by default, which is the same thing - that this function was using. But, using the define lets changes - take effect in this case, as well as the others where it was - already used. ........ - -2007-07-20 17:21 +0000 [r76055-76091] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r76087 | file | 2007-07-20 14:20:09 -0300 (Fri, - 20 Jul 2007) | 14 lines Merged revisions 76080 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 - lines (closes issue #10247) Reported by: fkasumovic Patches: - chan_sip.patch uploaded by fkasumovic (license #101) Drop any - peer realm authentication entries when reloading so multiple - entries do not get added to the peer. ........ ................ - - * /, res/res_convert.c: Merged revisions 76067 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r76067 | file | 2007-07-20 14:10:17 -0300 (Fri, 20 Jul 2007) | 6 - lines (closes issue #10246) Reported by: fkasumovic Patches: - res_conver.patch uploaded by fkasumovic (license #101) Use the - last occurance of . to find the extension, not the first - occurance. ........ - - * channels/chan_sip.c: It is impossible for the externhost variable - to not exist, it is however possible for it to be empty. - -2007-07-20 15:06 +0000 [r76034-76037] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: Don't use a field size for the last argument - of printf format, because in this case the string is left-aligned - and it is not truncated anyways. Omitting the field size prevents - the generation of trailing whitespace, which makes the string fit - in smaller windows. - - * channels/chan_sip.c: Extend the 'network settings' section with - indication on the localnet settings (requires the change in SVN - 76034), and also give an indication on whether/why/how the - remapping of addresses in SIP message is done or not. I think - this is especially useful for debugging the configuration, as the - address remapping depends on a combination of at least 3 - parameters (localnet, externhost, externip) and successful DNS - lookup. An example of the output of this section is below: - Network Settings: --------------------------- SIP address - remapping: Enabled using externhost Externhost: foo.dyndns.net - Externip: 80.64.128.23:0 Externrefresh: 10 Internal IP: - 12.34.56.78:5060 Localnet: 192.168.0.0/255.255.0.0 - 10.0.0.0/255.0.0.0 I leave to the community the judgement if the - above info is a useful addition for 1.4. It is not a bugfix, but - it is neither a new feature, only a useful diagnostic tool. Note - that I would like to move there also the bindaddress/port - information, in the usual addr:port format e.g. Bindaddress: - 0.0.0.0:5060 so that network information is all in one place. - - * include/asterisk/acl.h, main/acl.c: expose struct ast_ha so - external code can do things such as printing it (e.g. chan_sip.c - in a subsequent commit). Obviously exposing the internals of a - data structure is far from ideal (especially in a case like this - where the implementation is very inefficient and will need to be - changed at some point). On the other hand, it was also unclear - what additional APIs should we provide instead, and because - exposing the stucture has no impact on source and binary - compatibility, this seemed to me the best option at this time. - -2007-07-20 01:54 +0000 [r76015] Tilghman Lesher <tlesher@digium.com> - - * main/logger.c: Reduce some logging contention by switching - several locks over to rwlocks - -2007-07-19 23:24 +0000 [r75982-75983] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c, - channels/chan_sip.c, include/asterisk/dundi.h, - res/res_features.c, include/asterisk/chanspy.h, - include/asterisk/speech.h, channels/iax2-provision.c, - include/asterisk/cdr.h, include/asterisk/channel.h, - res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c, - channels/iax2-provision.h, main/loader.c, - include/asterisk/abstract_jb.h, include/asterisk/features.h, - main/channel.c, include/asterisk/app.h, funcs/func_odbc.c, - include/asterisk/module.h, include/asterisk/jabber.h, - apps/app_minivm.c, main/app.c, pbx/pbx_dundi.c, - apps/app_mixmonitor.c, apps/app_voicemail.c: After some study, - thought, comparing, etc. I've backed out the previous universal - mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit - version of ast_flags (ast_flags64), and 64-bit versions of the - test-flag, set-flag, etc. macros, and an app_parse_options64 - routine, and I use these in app_dial alone, to eliminate the - 30-option limit it had grown to meet. There is room now for 32 - more options and flags. I was heavily tempted to implement some - of the other ideas that were presented, but this solution does - not intro any new versions of dial, doesn't have a different API, - has a minimal/zero impact on code outside of dial, and doesn't - seriously (I hope) affect the code structure of dial. It's the - best I can think of right now. My goal was NOT to rewrite dial. I - leave that to a future, coordinated effort. - - * apps/app_queue.c: This repairs a 'warning: ISO C90 forbids mixed - declarations and code' message that cripples my dev-mode enabled - build - -2007-07-19 19:02 +0000 [r75977-75979] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 75978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75978 | mmichelson | 2007-07-19 13:59:30 -0500 (Thu, 19 Jul - 2007) | 3 lines The diff on this looks pretty big but all I did - was remove a pointless if statement (always evaluates true). - ........ - - * /, apps/app_queue.c: Merged revisions 75969 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul - 2007) | 10 lines Changes in handling return values of several - functions in app_queue. This all started as a fix for issue - #10008 but now includes all of the following changes: 1. - Simplifying the code to handle positive return values from ast - API calls. 2. Removing the background_file function. 3. The fix - for issue #10008 (closes issue #10008, reported and patched by - dimas) ........ - -2007-07-19 15:59 +0000 [r75911-75930] Russell Bryant <russell@digium.com> - - * res/res_agi.c: (closes issue #10210, reported and patched by - juggie) This merges the trunk only part of the patches from this - issue. In 1.4, res_agi will issue a warning if you try to use - DeadAGI on a channel that is not hung up. Now, in trunk, it just - plain won't let you do it. - - * /, channels/chan_iax2.c: Merged revisions 75928 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75928 | russell | 2007-07-19 10:53:15 -0500 - (Thu, 19 Jul 2007) | 14 lines Merged revisions 75927 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 - Jul 2007) | 6 lines When processing full frames, take sequence - number wraparound into account when deciding whether or not we - need to request retransmissions by sending a VNAK. This code - could cause VNAKs to be sent erroneously in some cases, and to - not be sent in other cases when it should have been. (closes - issue #10237, reported and patched by mihai) ........ - ................ - - * main/acl.c: Remove some debug code that was added in revision - 75894, which removed some other debug code. :) - -2007-07-19 12:38 +0000 [r75873-75894] Luigi Rizzo <rizzo@icir.org> - - * main/acl.c: comment out some terribly expensive debugging code in - the body of ast_apply_ha() - - * channels/chan_sip.c: print more of the network settings - (externip, externhost etc.) in the "sip show settings" cli - output. I have put these in a separate section, probably even - bindaddr and SIP port should go there. There are more things to - add here e.g. localnet and so on. - - * channels/chan_sip.c: document the use of externip, externhost and - other nat-related options, as well as the handling of the sip - socket. - - * channels/chan_sip.c: ast_sip_ouraddrfor() never fails, so make it - void and remove the code that would never be called. - - * channels/chan_sip.c: portability fix: use %f instead of %lf when - printing double. The l is useless. - -2007-07-19 04:45 +0000 [r75841-75857] Tilghman Lesher <tlesher@digium.com> - - * channels/misdn/ie.c, channels/misdn/isdn_lib.c: Allow chan_misdn - to build in dev-mode - - * apps/app_rpt.c: Fix trunk where I broke it earlier (for - ast_strftime branch) - -2007-07-18 23:00 +0000 [r75808] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75807 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r75807 | qwell | 2007-07-18 17:59:18 -0500 (Wed, 18 Jul - 2007) | 1 line Need to make sure we set milliseconds and - timestamp - pointed out by the recent ast_ time stuff from - Tilghman ........ - -2007-07-18 22:52 +0000 [r75806] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: I thought I noticed a memory leak earlier - when I saw that the contents of this list were not destroyed when - the module is unloaded. However, after reading the code related - to the use of this list a lot today, I realized that it isn't - necessary. So, I have added a comment to explain why it isn't - necessary. - -2007-07-18 22:40 +0000 [r75805] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Change IAX variables to use datastores - (closes issue #9315) - -2007-07-18 21:10 +0000 [r75761] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75759 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75759 | russell | 2007-07-18 16:09:46 -0500 - (Wed, 18 Jul 2007) | 13 lines Merged revisions 75757 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 - Jul 2007) | 5 lines When traversing the queue of frames for - possible retransmission after receiving a VNAK, handle sequence - number wraparound so that all frames that should be retransmitted - actually do get retransmitted. (issue #10227, reported and - patched by mihai) ........ ................ - -2007-07-18 20:43 +0000 [r75750] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 75749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75749 | tilghman | 2007-07-18 15:40:18 -0500 - (Wed, 18 Jul 2007) | 10 lines Merged revisions 75748 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 - Jul 2007) | 2 lines Store prior to copy (closes issue #10193) - ........ ................ - -2007-07-18 20:18 +0000 [r75714-75734] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75732 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul - 2007) | 1 line Umm, why are we transmitting dialtone on cfwdall? - ........ - - * /, channels/chan_skinny.c: Merged revisions 75711 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #9245) ........ r75711 | qwell | 2007-07-18 14:54:32 -0500 - (Wed, 18 Jul 2007) | 4 lines Fixes for 7935/7936 conference - phones. Issue 9245, patch by slimey. ........ - -2007-07-18 19:51 +0000 [r75710] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75707 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #9887) ........ r75707 | qwell | 2007-07-18 14:48:12 -0500 - (Wed, 18 Jul 2007) | 4 lines Fix issues with new 79x1 phones. - Issue 9887, patches by DEA ........ - -2007-07-18 19:50 +0000 [r75709] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: convert some lines indented with spaces to - tabs - -2007-07-18 19:47 +0000 [r75706] Tilghman Lesher <tlesher@digium.com> - - * main/say.c, funcs/func_strings.c, main/utils.c, - apps/app_alarmreceiver.c, include/asterisk/localtime.h, - cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c, - main/loader.c, main/cli.c, cdr/cdr_csv.c, main/cdr.c, - channels/chan_phone.c, main/manager.c, channels/chan_skinny.c, - cdr/cdr_sqlite.c, apps/app_minivm.c, channels/misdn/ie.c, - main/logger.c, main/http.c, main/stdtime/localtime.c, - cdr/cdr_odbc.c, apps/app_rpt.c, include/asterisk/options.h, - channels/chan_mgcp.c, cdr/cdr_manager.c, main/pbx.c, - channels/chan_zap.c, funcs/func_timeout.c, channels/chan_sip.c, - channels/chan_agent.c, channels/iax2-parser.c, - apps/app_playback.c, cdr/cdr_tds.c, main/callerid.c, - res/snmp/agent.c, apps/app_sms.c, include/asterisk/strings.h, - main/asterisk.c, apps/app_voicemail.c: Merge in ast_strftime - branch, which changes timestamps to be accurate to the - microsecond, instead of only to the second - -2007-07-18 17:59 +0000 [r75659] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, apps/app_queue.c: Merged revisions 75658 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75658 | dhubbard | 2007-07-18 12:56:30 -0500 - (Wed, 18 Jul 2007) | 9 lines Merged revisions 75657 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 - Jul 2007) | 1 line removed the word 'pissed' from ast_log(...) - function call for BE-90 ........ ................ - -2007-07-18 15:45 +0000 [r75586-75624] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 75623 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 - lines Few more places that needs to check for onhold state. - ........ - - * /, channels/chan_sip.c: Merged revisions 75621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 - lines (closes issue #10165) Reported by: elandivar It is possible - for hold status to exist without call limits set, so we need to - ensure update_call_counter is executed regardless. ........ - - * /, channels/chan_h323.c: Merged revisions 75619 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2 - lines Don't bother reloading chan_h323 if it did not load - successfully in the first place. This would otherwise cause a - crash. ........ - - * funcs/func_curl.c: Clean up func_curl a bit. - -2007-07-18 14:35 +0000 [r75585] Steve Murphy <murf@digium.com> - - * main/channel.c, channels/chan_sip.c, res/res_features.c, - pbx/pbx_dundi.c, main/rtp.c, apps/app_voicemail.c: This corrects - the problem with flags and %lld formats on 64-bit machines, where - uint64_t is NOT acceptable for %lld, and also works on 32-bit - machines. At least, with gcc. - -2007-07-18 14:20 +0000 [r75566-75584] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 75583 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75583 | file | 2007-07-18 11:18:53 -0300 (Wed, 18 Jul 2007) | 5 - lines (closes issue #10224) Reported by: irroot Record the - threadid of each running thread before shutting them down as the - thread themselves may change the value. ........ - - * channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_realtime.c, - apps/app_voicemail.c: Minor code tweaks. Variables were being - checked wrong in some situations and didn't need to be checked in - others. - -2007-07-18 12:38 +0000 [r75530] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_meetme.c: Merged revisions 75529 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75529 | tilghman | 2007-07-18 07:29:41 -0500 (Wed, 18 Jul 2007) - | 2 lines Using a freed frame causes crashes (closes issue #9317) - ........ - -2007-07-17 21:52 +0000 [r75505] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Spotted this bug today myself, trying to reproduce - a BE bug. Use a vert bar instead of a comma, when calling RAND. - -2007-07-17 20:58 +0000 [r75446-75451] Russell Bryant <russell@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75450 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75450 | russell | 2007-07-17 15:57:56 -0500 - (Tue, 17 Jul 2007) | 11 lines Merged revisions 75449 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 - Jul 2007) | 3 lines Properly check for the length in the skinny - packet to prevent an invalid memcpy. (ASA-2007-016) ........ - ................ - - * channels/iax2-parser.h, /, channels/chan_iax2.c, - channels/iax2-parser.c: Merged revisions 75445 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75445 | russell | 2007-07-17 15:48:21 -0500 - (Tue, 17 Jul 2007) | 13 lines Merged revisions 75444 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 - Jul 2007) | 5 lines Ensure that when encoding the contents of an - ast_frame into an iax_frame, that the size of the destination - buffer is known in the iax_frame so that code won't write past - the end of the allocated buffer when sending outgoing frames. - (ASA-2007-014) ........ ................ - -2007-07-17 20:42 +0000 [r75438-75442] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75441 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75441 | russell | 2007-07-17 15:42:12 -0500 - (Tue, 17 Jul 2007) | 12 lines Merged revisions 75440 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 - Jul 2007) | 4 lines After parsing information elements in IAX - frames, set the data length to zero, so that code later on does - not think it has data to copy. (ASA-2007-015) ........ - ................ - -2007-07-17 20:05 +0000 [r75406] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /: Merged revisions 75405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul - 2007) | 6 lines Fixing an error I made earlier. ast_fileexists - can return -1 on failure, so I need to be sure that we only enter - the if statement if it is successful. Related to my fix to issue - #10186 ........ - -2007-07-17 20:01 +0000 [r75402-75404] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 75403 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) | - 12 lines (closes issue #10209) Reported by: juggie Patches: - 10209-trunk-2.patch uploaded by juggie Tested by: juggie, - blitzrage In ast_pbx_run(), mark a channel as hung up after an - application returned -1, or when it runs out of extensions to - execute. This is so that code can detect that this channel has - been hung up for things like making sure DeadAGI is used on - actual dead channels, and is beneficial for other things, like - making sure someone doesn't try to start spying on a channel that - is about to go away. ........ - - * /, res/res_agi.c: Merged revisions 75401 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75401 | russell | 2007-07-17 14:45:07 -0500 (Tue, 17 Jul 2007) | - 3 lines Remove a duplicated newline character in AGI debug - output. (closes issue #10207, patch by seanbright) ........ - -2007-07-17 19:40 +0000 [r75400] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c, - channels/chan_sip.c, include/asterisk/dundi.h, - res/res_features.c, include/asterisk/chanspy.h, - include/asterisk/speech.h, channels/iax2-provision.c, - include/asterisk/cdr.h, include/asterisk/channel.h, - res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c, - channels/iax2-provision.h, main/loader.c, - include/asterisk/features.h, include/asterisk/abstract_jb.h, - main/channel.c, funcs/func_odbc.c, include/asterisk/module.h, - include/asterisk/jabber.h, apps/app_minivm.c, utils/ael_main.c, - pbx/pbx_dundi.c, apps/app_mixmonitor.c, utils/check_expr.c, - apps/app_voicemail.c: via 10206, I have added an option (e) to - Dial to allow the h exten to get run on peer. Had to upgrade - ast_flag stuff to 64 bits to do this. - -2007-07-17 14:48 +0000 [r75381] Joshua Colp <jcolp@digium.com> - - * include/asterisk/config.h: Make trunk build once again. - -2007-07-17 14:32 +0000 [r75365-75379] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/config.h, main/config.c: Introduce - ast_parse_arg() , a generic function to parse strings in a - consistent way. This is meant to replace the custom code which is - repeated all over the place in the various files when parsing - config files, CLI entries and other string information. Right now - the code supports parsing int32, uint32 and sockaddr_in with - optional default values and bound checks. It contains minimal - error checking, but that can be easily extended as the need - arises. Being a new API i am introducing this only in trunk, - though I believe that once the interface has been ironed out it - might become a worthwhile addition to 1.4 as well - basically, - the first time we will need to fix a piece of argument parsing - code, we might as well bring in this change and use the new API - instead. - - * apps/app_minivm.c: Initialize a variable to avoid a warning when - the compiler (and/or the optimization level) may think it is used - uninitialized. The code was indeed correct, but unfortunately the - result of some compiler checks such as -Wunused and - -Wuninitialized depends heavily on the optimization level. - -2007-07-17 12:01 +0000 [r75351] Jason Parker <jparker@digium.com> - - * apps/app_dial.c: Fix an incorrect parenthesization (TODO: Find a - better word) in app_dial Pointed out by Fanzhou Zhao Closes issue - #10216 - -2007-07-16 20:58 +0000 [r75307] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/dns.c: Merged revisions 75306 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75306 | kpfleming | 2007-07-16 15:53:24 -0500 - (Mon, 16 Jul 2007) | 11 lines Merged revisions 75304 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 - Jul 2007) | 3 lines provide proper copyright/license attribution - for this structure that was copied from a BSD-licensed header - file long, long ago... ........ ................ - -2007-07-16 18:38 +0000 [r75255-75260] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, include/asterisk/pbx.h: Change the function name - slightly... just for kpfleming! - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Add in - check for the GCC attribute deprecated. It may be used soon! - - * funcs/func_enum.c, funcs/func_rand.c, main/pbx.c, - funcs/func_curl.c, funcs/func_version.c, funcs/func_cut.c, - funcs/func_vmcount.c, include/asterisk/pbx.h, - funcs/func_realtime.c: For my next trick I will make it so - dialplan functions no longer need to call ast_module_user_add and - ast_module_user_remove. These are now called in the ast_func_read - and ast_func_write functions outside of the module. - -2007-07-16 18:18 +0000 [r75254] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /: Merged revisions 75253 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul - 2007) | 8 lines Restoring functionality from 1.2 wherein - Retrydial will not exit if there is no announce file specified. - This change makes it so that if there is no announce file - specified, the application will continue until finished (or - caller hangs up). If a bogus announce file is specified, then a - warning message will be printed saying that the file could not be - found, but execution will still continue. (closes issue #10186, - reported by jon, patched by me) ........ - -2007-07-16 15:57 +0000 [r75183-75227] Joshua Colp <jcolp@digium.com> - - * apps/app_verbose.c: I found this sillyness when I did my - ast_module_user conversion. Return immediately if no data was - passed to the Verbose application. - - * apps/app_readfile.c, apps/app_record.c, apps/app_sayunixtime.c, - apps/app_test.c, apps/app_alarmreceiver.c, apps/app_image.c, - apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, - apps/app_skel.c, apps/app_zapscan.c, apps/app_dumpchan.c, - apps/app_zapras.c, apps/app_amd.c, apps/app_url.c, - apps/app_externalivr.c, apps/app_milliwatt.c, apps/app_dial.c, - main/pbx.c, apps/app_page.c, apps/app_privacy.c, apps/app_echo.c, - apps/app_softhangup.c, apps/app_disa.c, apps/app_morsecode.c, - apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, - apps/app_playback.c, apps/app_speech_utils.c, - apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, - apps/app_macro.c, apps/app_zapateller.c, apps/app_chanspy.c, - apps/app_mixmonitor.c, apps/app_cdr.c, apps/app_voicemail.c, - apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c, - apps/app_userevent.c, apps/app_followme.c, - apps/app_controlplayback.c, apps/app_osplookup.c, - apps/app_setcallerid.c, apps/app_minivm.c, apps/app_mp3.c, - apps/app_directory.c, apps/app_rpt.c, apps/app_ivrdemo.c, - apps/app_parkandannounce.c, apps/app_adsiprog.c, - apps/app_while.c, apps/app_nbscat.c, apps/app_read.c, - apps/app_festival.c, apps/app_system.c, apps/app_getcpeid.c, - apps/app_queue.c, apps/app_channelredirect.c, apps/app_forkcdr.c, - apps/app_flash.c, apps/app_directed_pickup.c, apps/app_sms.c, - include/asterisk/pbx.h, apps/app_senddtmf.c, apps/app_stack.c, - apps/app_verbose.c: Applications no longer need to call - ast_module_user_add and ast_module_user_remove. This is now taken - care of in the pbx_exec function outside of the application. - - * apps/app_readfile.c, res/res_features.c, apps/app_record.c, - apps/app_sayunixtime.c, apps/app_test.c, - apps/app_alarmreceiver.c, apps/app_image.c, - apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, - apps/app_zapscan.c, apps/app_dumpchan.c, apps/app_zapras.c, - apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c, - apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, - apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c, - apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c, - apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c, - apps/app_speech_utils.c, funcs/func_curl.c, apps/app_zapbarge.c, - apps/app_waitforring.c, apps/app_sendtext.c, apps/app_macro.c, - apps/app_zapateller.c, apps/app_mixmonitor.c, apps/app_chanspy.c, - apps/app_cdr.c, apps/app_voicemail.c, apps/app_meetme.c, - apps/app_authenticate.c, apps/app_userevent.c, - funcs/func_vmcount.c, apps/app_followme.c, funcs/func_enum.c, - res/res_config_odbc.c, apps/app_setcallerid.c, - apps/app_osplookup.c, apps/app_minivm.c, res/res_agi.c, - apps/app_mp3.c, res/res_realtime.c, apps/app_rpt.c, - apps/app_ivrdemo.c, apps/app_parkandannounce.c, - apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c, - res/res_config_pgsql.c, apps/app_read.c, apps/app_festival.c, - apps/app_waitforsilence.c, apps/app_system.c, apps/app_queue.c, - apps/app_getcpeid.c, funcs/func_realtime.c, apps/app_forkcdr.c, - apps/app_channelredirect.c, apps/app_flash.c, - funcs/func_blacklist.c, apps/app_sms.c, apps/app_senddtmf.c, - apps/app_stack.c, apps/app_verbose.c: It is no longer required - for each module that deals with a channel to call - ast_module_user_hangup_all in it's unload function. The loader - will automatically perform this action for it. - -2007-07-16 02:51 +0000 [r75163-75164] Russell Bryant <russell@digium.com> - - * include/asterisk/devicestate.h, include/asterisk/dundi.h, - include/asterisk/enum.h, include/asterisk/config.h, - include/asterisk/io.h, include/asterisk/cli.h, - include/asterisk/channel.h, include/asterisk/cdr.h, - include/asterisk/manager.h, include/asterisk/tdd.h, - include/asterisk/abstract_jb.h, include/asterisk/file.h, - include/asterisk/res_odbc.h, include/asterisk/adsi.h, - include/asterisk/crypto.h, include/asterisk/doxyref.h, - include/asterisk/image.h, include/asterisk/musiconhold.h, - include/asterisk/jabber.h, include/asterisk/linkedlists.h, - include/asterisk/module.h, include/asterisk/strings.h, - include/asterisk/pbx.h, include/asterisk/frame.h, - include/asterisk/say.h, include/asterisk/translate.h: Merge a - bunch of doxygen updates to header files. This includes changes - to use the \retval tag for documenting return values, fixing - various warnings when generating the documentation, and various - other things. (closes issue #10203, snuffy) - - * funcs/func_iconv.c: Cast the 2nd argument to iconv() to a void *, - as some systems define it as a (const char *), while others - define it as (char *). This is done to suppress compiler warnings - about it. - -2007-07-13 20:37 +0000 [r75109] Russell Bryant <russell@digium.com> - - * /: Merged revisions 75108 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75108 | russell | 2007-07-13 15:36:16 -0500 - (Fri, 13 Jul 2007) | 11 lines Merged revisions 75107 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 - Jul 2007) | 3 lines Fix a couple potential minor memory leaks. - load_moh_classes() could return without destroying the loaded - configuration. ........ ................ - -2007-07-13 20:16 +0000 [r75082] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 75078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75078 | mmichelson | 2007-07-13 15:15:30 -0500 - (Fri, 13 Jul 2007) | 13 lines Merged revisions 75066 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 - Jul 2007) | 5 lines Fixed an issue where chanspy flags were - uninitialized if no options were passed. What triggered this - investigation was an IRC chat where some people's quiet flags - were set while others' weren't even though none of them had - specified the q option. ........ ................ - -2007-07-13 20:15 +0000 [r75054-75077] Russell Bryant <russell@digium.com> - - * main/rtp.c: resolve a compiler warning - - * /, res/res_musiconhold.c: Merged revisions 75067 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75067 | russell | 2007-07-13 15:10:40 -0500 - (Fri, 13 Jul 2007) | 14 lines Merged revisions 75059 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 - Jul 2007) | 6 lines Ensure that adding a user to the list of - users of a specific music on hold class is not done at the same - time as any of the other operations on this list to prevent list - corruption. Using the global moh_data lock for this is not ideal, - but it is what is used to protect these lists everywhere else in - the module, and I am only changing what is necessary to fix the - bug. ........ ................ - - * channels/chan_zap.c, /: Merged revisions 75053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r75053 | russell | 2007-07-13 14:11:26 -0500 - (Fri, 13 Jul 2007) | 20 lines Merged revisions 75052 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 - Jul 2007) | 12 lines (closes issue #9660) Reported by: mmacvicar - Patches submitted by: bbryant, russell Tested by: mmacvicar, - marco, arcivanov, jmhunter, explidous When using a TDM400P (and - probably other analog cards) there was a chance that you could - hang up and pick the phone back up where it has been long enough - to be not considered a flash hook, but too soon such that the - device reports that it is busy and the person on the phone will - only hear silence. This patch makes chan_zap more tolerant of - this and gives the device a couple of seconds to succeed so the - person on the phone happily gets their dialtone. ........ - ................ - -2007-07-13 16:22 +0000 [r75034] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/rtp.h, main/rtp.c: Small improvement to the STUN - support so it can be used by sockets other than RTP ones. The - main change is a new API function in main/rtp.c (see there for a - description) int ast_stun_request(int s, struct sockaddr_in *dst, - const char *username, struct sockaddr_in *answer) which can be - used to send an STUN request on a socket, and optionally wait for - a reply and store the STUN_MAPPED_ADDRESS into the 'answer' - argument (obviously, the version that waits for a reply is - blocking, but this is no different from DNS resolutions). - Internally there are minor modifications to let - stun_handle_packet() be somewhat configurable on how to parse the - body of responses. At the moment i am not committing any change - to the clients, but adding STUN client support is extremely - simple, e.g. chan_sip.c could do something like this: + add a - variable to store the stun server address; static struct - sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add - code to parse a config file of the form - "stunaddr=my.stun.server.org:3478" (not shown for brevity); + - right after binding the main sip socket, talk to the stun server - to determine the externally visible address if - (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock, - &stunaddr, NULL, &externip); so now 'externip' is set with the - externally visible address. so it is really trivial. Similarly - ast_stun_request could be called when creating the RTP socket - (possibly adding a struct sockaddr_in field in the struct ast_rtp - to store the externalip). - -2007-07-12 23:02 +0000 [r74999] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 74997 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ ........ - -2007-07-12 20:46 +0000 [r74956] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 74955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 - line This patch resolves 10143; thanks to irroot for the patch; - looked acceptable. Let the community decide if it messes things - up ........ - -2007-07-12 19:19 +0000 [r74891-74923] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 74922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2 - lines Whoops... didn't want this to be returned to 0 each - iteration. ........ - - * main/channel.c, /: Merged revisions 74888 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2 - lines When waiting for a digit ensure that a begin frame was - received with it, not just an end frame. (issue #10084 reported - by rushowr) ........ - -2007-07-12 16:54 +0000 [r74865-74867] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 74866 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul - 2007) | 1 line It helps if I actually add this stuff for the 7921 - too - otherwise it won't actually do much of anything. ........ - - * /, channels/chan_skinny.c: Merged revisions 74864 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul - 2007) | 1 line Add device ID for 7921 wireless skinny phone - ........ - -2007-07-12 16:21 +0000 [r74850] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: more cleanup, this time to stun_handle_packet(). - Among other things: + mark a potentially dangerous - write-past-end-of-buffer + localize some variables in the block - generating stun replies. As before, not ready yet for a merge to - 1.4 - -2007-07-12 15:55 +0000 [r74816] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 74815 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74815 | file | 2007-07-12 12:53:55 -0300 (Thu, - 12 Jul 2007) | 10 lines Merged revisions 74814 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul 2007) | 2 - lines Only print out a warning for situations where it is - actually helpful. (issue #10187 reported by denke) ........ - ................ - -2007-07-12 15:42 +0000 [r74813] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: a little bit of code cleanup to rtp.c, mostly to - function ast_rtp_new_with_bindaddr(): 1. add comments to the - logic of the main loop; 2. use a common exit point on failure so - the cleanup is done only in one place; 3. handle failures in - rtp_socket() in the main loop of the function; No functional - changes except for #3 above, so it is not yet worthwhile merging - this and other changes to 1.4 Once the cleanup work on this file - will be complete (which among other things should include some - extensions to the stun support) it might be a good thing to push - all the changes to 1.4 - -2007-07-11 23:05 +0000 [r74769] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 74767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74767 | russell | 2007-07-11 17:57:07 -0500 - (Wed, 11 Jul 2007) | 13 lines Merged revisions 74766 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 - Jul 2007) | 5 lines The function make_trunk() can fail and return - -1 instead of a valid new call number. Fix the uses of this - function to handle this instead of treating it as the new call - number. This would cause a deadlock and memory corruption. - (possible cause of issue #9614 and others, patch by me) ........ - ................ - -2007-07-11 21:15 +0000 [r74726] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 74722 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74722 | mmichelson | 2007-07-11 16:14:09 -0500 - (Wed, 11 Jul 2007) | 13 lines Merged revisions 74719 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 - Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" - did not work...at all. Now it does. (closes issue #10178, - reported and patched by makoto, with slight modification for 1.4 - and trunk by me) ........ ................ - -2007-07-11 21:09 +0000 [r74703-74713] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Code cleanup of res_agi - - * res/res_smdi.c: Code cleanup of res_smdi - - * pbx/pbx_spool.c: Clean up pbx_spool. So many nested if - statements... - - * main/udptl.c, include/asterisk/udptl.h: Use linkedlist macros for - UDPTL protocol list. - -2007-07-11 18:35 +0000 [r74658] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c: Merged revisions 74657 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74657 | russell | 2007-07-11 13:34:51 -0500 - (Wed, 11 Jul 2007) | 12 lines Merged revisions 74656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 - Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows - the condition that uses LIKE. This fixes realtime extensions with - ODBC. (closes issue #10175, reported by stuarth, patch by me) - ........ ................ - -2007-07-11 18:21 +0000 [r74636-74648] Steve Murphy <murf@digium.com> - - * Makefile, /: Merged revisions 74642 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74642 | murf | 2007-07-11 12:18:42 -0600 (Wed, 11 Jul 2007) | 1 - line This fixes 10172, where the entire man8 dir gets removed - during an uninstall of asterisk ........ - - * /: blocking 74628 from trunk... only applied to 1.4 - -2007-07-11 17:34 +0000 [r74575-74616] Joshua Colp <jcolp@digium.com> - - * include/asterisk/speech.h, res/res_speech.c, - apps/app_speech_utils.c: Use the linkedlists.h AST_LIST_NEXT - macro for modifying the list of results. - - * channels/chan_phone.c, /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 74572 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2 - lines Instead of figuring out kernel versions that have - compiler.h and not... let's just use autoconf to check for it's - presence. (issue #10174 reported by francesco_r) ........ - -2007-07-11 16:24 +0000 [r74571] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: add a bit of documentation on what the stun code in - rtp.c does (which is very little, at the moment). Eventually, - when the functionality is extended, the changes can be merged - back to 1.4. At the moment this is pointless. Note, this change - is whitespace only. - -2007-07-11 16:19 +0000 [r74516-74570] Joshua Colp <jcolp@digium.com> - - * include/asterisk/speech.h, res/res_speech.c, - apps/app_speech_utils.c: Allow the native formats of a channel to - influence the audio that is going to the engine. The best format - will try to be chosen with an ultimate fallback to signed linear - if possible. - - * res/res_speech.c: Can't forget to remember what format is in use - for writing. - - * include/asterisk/speech.h, res/res_speech.c: Change the speech - API to allow passing the format through to the engine. - - * channels/misdn/isdn_lib_intern.h: Change header a bit to get rid - of a doxygen parse error. (issue #10177 reported by snuffy) - - * channels/chan_phone.c, /: Merged revisions 74515 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul - 2007) | 2 lines Only check if we need to do a SIGMA based tone - generation if we have a card. (issue #10179 reported by mikowhy) - ........ - -2007-07-10 23:34 +0000 [r74477] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 74476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74476 | mmichelson | 2007-07-10 18:32:52 -0500 (Tue, 10 Jul - 2007) | 5 lines Forwarding a message with IMAP storage was - storing the message in the sender's box instead of the forwarded - mailbox. (closes issue #10138, reported and patched by jaroth) - ........ - -2007-07-10 20:02 +0000 [r74375-74429] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 74428 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10158) ................ r74428 | qwell | 2007-07-10 - 14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines Merged revisions - 74427 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 - lines Fix an issue where it was possible to have a service level - of over 100% Between the time recalc_holdtime and update_queue - was called, it was possible that the call could have been hungup. - Move both additions to the same place, so this won't happen. - Issue 10158, initial patch by makoto, modified by me. ........ - ................ - - * /, main/dns.c: Merged revisions 74388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74388 | qwell | 2007-07-10 14:10:36 -0500 (Tue, 10 Jul 2007) | 4 - lines Don't use #if to check if something is defined - use #ifdef - instead. Pointed out by kpfleming ........ - - * /, channels/chan_agent.c: Merged revisions 74379 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10169) ................ r74379 | qwell | 2007-07-10 - 14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions - 74376 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 - lines Fix an issue with wrapuptime not working when using - AgentLogin. Issue 10169, patch by makoto, with a minor mod by me - to not re-break issue 9618 ........ ................ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - main/dns.c: Merged revisions 74374 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes - issue #10133) ................ r74374 | qwell | 2007-07-10 - 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines Merged revisions - 74373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 - lines Use res_ndestroy on systems that have it. Otherwise, use - res_nclose. This prevents a memleak on NetBSD - and possibly - others. Issue 10133, patch by me, reported and tested by scw - ........ ................ - -2007-07-10 16:01 +0000 [r74324] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 74323 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74323 | russell | 2007-07-10 11:00:11 -0500 (Tue, 10 - Jul 2007) | 1 line fix an uninitialized variable ........ - -2007-07-10 15:41 +0000 [r74318-74319] Jason Parker <jparker@digium.com> - - * /: svn revert != svn resolved Fix merged property... - - * apps/app_voicemail.c: Merged revisions 74317 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes - issue #10170) ................ r74317 | qwell | 2007-07-10 - 10:38:32 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions - 74316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 - lines Fix a small typo in description in of Voicemail() - application. Issue 10170, patch by casper. ........ - ................ - -2007-07-10 15:32 +0000 [r74315] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 74314 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74314 | russell | 2007-07-10 10:31:41 -0500 - (Tue, 10 Jul 2007) | 11 lines Merged revisions 74313 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 - Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue - #10075, this part reported by jmls on IRC, patch by me) ........ - ................ - -2007-07-10 15:07 +0000 [r74272] Jason Parker <jparker@digium.com> - - * channels/chan_agent.c, include/asterisk/monitor.h, - apps/app_queue.c, res/res_monitor.c: Fix building that was broken - by recent monitor.h changes. Thanks Russell for pointing this out - (and pointing out what I probably did to prevent gcc from fixing - it - don't ctrl-C builds) - -2007-07-10 14:51 +0000 [r74263-74266] Joshua Colp <jcolp@digium.com> - - * /, main/app.c: Merged revisions 74265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r74265 | file | 2007-07-10 11:50:00 -0300 (Tue, - 10 Jul 2007) | 10 lines Merged revisions 74264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 - lines Ensure the group information category exists before trying - to do a string comparison with it. (issue #10171 reported by - mlegas) ........ ................ - -2007-07-09 21:32 +0000 [r74212] Russell Bryant <russell@digium.com> - - * /, configure, configure.ac: Merged revisions 74211 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74211 | russell | 2007-07-09 16:31:30 -0500 (Mon, 09 - Jul 2007) | 5 lines Update the configure script to check for a - required function that is not present in the 1.2 version of - libpri. This will prevent the configure script from thinking that - it has compatible libpri support for Asterisk 1.4, when it - actually does not because the installed version is from 1.2. - ........ - -2007-07-09 20:58 +0000 [r74164] Jason Parker <jparker@digium.com> - - * include/asterisk/monitor.h, res/res_monitor.c: (closes issue - #7596) Reported by: julien23 Patches submitted by: julien23 Add - the ability to disable recording the input or output streams in - res_monitor. - -2007-07-09 20:54 +0000 [r74163] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 74162 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09 - Jul 2007) | 9 lines (closes issue #10123) Reported by: blitzrage - Patches submitted by: juggie, qwell, me Tested by: blitzrage When - trying to find a music on hold class to use, try all of the - options, instead of only the first one that is set. Also, change - the MusicOnHold applications to not hang up on the channel when a - class can not be found. ........ - -2007-07-09 20:21 +0000 [r74160] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 74159 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 Closes issue - #9186 ................ r74159 | qwell | 2007-07-09 15:19:28 -0500 - (Mon, 09 Jul 2007) | 16 lines Merged revisions 74158 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul - 2007) | 8 lines Several chan_zap options were not working on - reload because they were arbitrarily disallowed when reloading - some/most PRI options (such as signalling) was disallowed. - Options such as polarityonanswerdelay and answeronpolarityswitch - can safely be changed on a reload. This corrects that behavior. - Issue 9186, patch by tzafrir. ........ ................ - -2007-07-09 18:58 +0000 [r74125] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: remove an unused variable - -2007-07-09 18:43 +0000 [r74121-74123] Mark Michelson <mmichelson@digium.com> - - * /: Merged revisions 74122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74122 | mmichelson | 2007-07-09 13:38:28 -0500 (Mon, 09 Jul - 2007) | 3 lines Forgot to get rid of an extraneous debug message. - ........ - - * /, apps/app_queue.c: Merged revisions 74120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul - 2007) | 6 lines The n option for Queue should make the queue exit - immediately after failure to reach any members and should not be - dependent on the timeout value passed to Queue (closes issue - #10127, reported by bcnit, repaired by me) ........ - -2007-07-09 16:35 +0000 [r74084] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Add Queue and DestinationChannel headers to the - AgentCalled manager event to be more like the rest of the events - in this module. (closes issue #10114, patch by kwakwaversal) - -2007-07-09 15:34 +0000 [r74083] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 74082 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul - 2007) | 2 lines Only destroy the scheduler context if it was - allocated. (issue #10124 reported by gzero) ........ - -2007-07-09 14:58 +0000 [r74048] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 74047 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74047 | mmichelson | 2007-07-09 09:57:41 -0500 (Mon, 09 Jul - 2007) | 4 lines Fixed a logic error in leave_voicemail. Pass the - mailbox instead of the context to inbox_count when the context is - "default." (closes issue #10135, reported by yannj, repaired by - me) ........ - -2007-07-09 14:50 +0000 [r74044-74046] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_skinny.c, pbx/pbx_dundi.c: Merged revisions - 74045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2 - lines Few minor thread synchronization tweaks. (issue #10124 - reported by gzero) ........ - - * /, configure, acinclude.m4: Merged revisions 74043 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r74043 | file | 2007-07-09 11:34:33 -0300 (Mon, 09 Jul - 2007) | 2 lines Use AC_CHECK_HEADER to check for ptlib/openh323 - to allow for cross compiling. (issue #9675 reported by zandbelt) - ........ - -2007-07-09 08:30 +0000 [r74024-74025] Olle Johansson <oej@edvina.net> - - * CHANGES: Update with new features - - * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, - include/asterisk/channel.h: Implementation of a feature that will - disable "missed calls" counters on SIP phones. If the call is - answered by another phone, other phones won't display the call as - "missed". You can also add an option to the dial command so that - you can have a "followme" scenario and not count the calls as - "missed" when you cancel the call. Thanks to Ramon and Frank for - feedback on this feature. - -2007-07-09 04:09 +0000 [r73994] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/app.h, /, channels/chan_sip.c, - main/ast_expr2f.c, include/asterisk/channel.h, - funcs/func_devstate.c, apps/app_voicemail.c: Merged revisions - 73985 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) - | 2 lines Doxygen formatting fixes; fixes errors while 'make - progdocs'. (Closes issue #10104) ........ - -2007-07-09 03:14 +0000 [r73931-73983] Joshua Colp <jcolp@digium.com> - - * main/cdr.c, /: Merged revisions 73980 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73980 | file | 2007-07-09 00:13:19 -0300 (Mon, 09 Jul 2007) | 2 - lines Give Agent channel names priority when doing CDR merging. - (issue #10011 reported by krtorio) ........ - - * res/res_features.c: Use linkedlist macros for parking. - - * main/manager.c: Make sure the idText variable is empty, and put - it in the right place for the manager ack packet. (issue #10152 - reported by srt) - - * /, pbx/pbx_config.c: Merged revisions 73930 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73930 | file | 2007-07-08 22:13:57 -0300 (Sun, 08 Jul 2007) | 2 - lines Add a few sanity checks when writing out the dialplan. - (issue #10157 reported by dome) ........ - -2007-07-08 21:01 +0000 [r73911] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, - main/ast_expr2.y, configure.ac, main/ast_expr2.c: Restore EXP2 - and LOG2 functions, by providing mathematical identify functions, - when the underlying C functions are not available. - -2007-07-08 13:22 +0000 [r73886] Russell Bryant <russell@digium.com> - - * res/res_features.c: ast_exists_extension() does not return an - ast_device_state, so change this function to explicitly check for - the int return value. Also, make a few other minor changes such - as removing a variable. - -2007-07-08 09:49 +0000 [r73850] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 73849 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 - lines While tracking down a bug, I need some more history. - Dumphistory is very useful, indeed. ........ - -2007-07-07 16:44 +0000 [r73821] Steve Murphy <murf@digium.com> - - * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.y, - configure.ac, bootstrap.sh, main/ast_expr2.c: These changes fix - 10145 and 10150, a prob with BSD and exp2/log2 not existing, as - well as the bootstrap needing a small upgrade for openbsd. Many - thanks to mvanbaak - -2007-07-06 23:05 +0000 [r73771] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73769 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73769 | russell | 2007-07-06 18:02:58 -0500 - (Fri, 06 Jul 2007) | 12 lines Merged revisions 73768 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 - Jul 2007) | 4 lines If a sip_pvt struct has already registered an - extension state callback, remove the old one before adding a new - one. If this isn't done, Asterisk will crash. (issue #10120) - ........ ................ - -2007-07-06 16:39 +0000 [r73728] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 73727 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73727 | mmichelson | 2007-07-06 11:36:17 -0500 (Fri, 06 Jul - 2007) | 8 lines Fixing a rare case which causes voicemail to - crash when compiled with IMAP storage. inboxcount has the - possibility of finding an "interactive" vm_state when no - persistent "non-interactive" vm_state exists for that mailbox. If - this should happen when someone attempts to leave a message, it - results in a crash. This patch, along with my commit in revision - 72670 fix issue 10053, reported by jaroth. closes issue #10053 - ........ - -2007-07-06 16:30 +0000 [r73726] Kevin P. Fleming <kpfleming@digium.com> - - * main/minimime/mimeparser.yy.c, main/minimime/mimeparser.h, - main/minimime/mimeparser.tab.c, main/minimime/mimeparser.y, - main/minimime/Makefile, main/minimime/mimeparser.l, - main/minimime/mimeparser.tab.h, main/minimime/mm_parse.c: - eliminate another batch of compiler warnings (and a bug, although - in code we aren't using)... note that this required manually - editing the lexer output code (generated by flex), so some of - them will come back if the lexer is rebuilt - -2007-07-06 16:14 +0000 [r73680-73701] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 73696 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73696 | russell | 2007-07-06 11:12:51 -0500 - (Fri, 06 Jul 2007) | 16 lines Merged revisions 73684 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 - Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras - Patches submitted by: Corydon76 Tested by: apsaras Fix a problem - with MSSQL 2005 by explicitly stating that '\' is being used as - an escape character. ........ ................ - - * /, channels/chan_sip.c: Merged revisions 73679 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73679 | russell | 2007-07-06 10:57:25 -0500 - (Fri, 06 Jul 2007) | 15 lines Merged revisions 73678 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 - Jul 2007) | 7 lines (closes issue #10125) Reported by: makoto - Patches submitted by: makoto This fixes a crash in chan_sip that - happens when the bindaddr setting is not valid on Asterisk - startup, gets fixed, and then a reload gets issued. ........ - ................ - -2007-07-06 15:47 +0000 [r73677] Kevin P. Fleming <kpfleming@digium.com> - - * channels/busy.h (added), channels/ringtone.h (added), - channels/Makefile, channels: it really seems pointless to run - gentone to create these header files every time we build - Asterisk... - -2007-07-06 15:28 +0000 [r73676] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 73675 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73675 | mmichelson | 2007-07-06 10:27:28 -0500 - (Fri, 06 Jul 2007) | 13 lines Merged revisions 73674 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 - Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. - (issue 9618, reported by jiddings, patched by moi) closes issue - #9618 ........ ................ - -2007-07-06 03:48 +0000 [r73557-73633] Russell Bryant <russell@digium.com> - - * CHANGES: Redistribute a lot of the items that were in the Misc. - section - - * CHANGES: note TLS support for manager and HTTP in CHANGES - - * CREDITS: Philippe was listed twice - - * /, BUGS: Merged revisions 73629 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73629 | russell | 2007-07-05 22:34:46 -0500 (Thu, 05 Jul 2007) | - 1 line fix a little spelling error ........ - - * /, channels/chan_sip.c: Merged revisions 73598 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) | - 3 lines Fix a crash in chan_sip. Don't try to stop the monitor - thread if it was never started. (closes issue #10124, reported by - gzero, fixed by me) ........ - - * /, channels/chan_iax2.c: Merged revisions 73555 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73555 | russell | 2007-07-05 18:05:33 -0500 (Thu, 05 Jul 2007) | - 3 lines copy from the correct buffer when deferring a full frame - (related to issue #9937) ........ - -2007-07-05 22:48 +0000 [r73553] Kevin P. Fleming <kpfleming@digium.com> - - * main/minimime/mm_contenttype.c, main/minimime/mm_envelope.c, - main/minimime/mm_mimepart.c, main/minimime/mm_param.c, - main/minimime/mm_context.c, main/minimime/mm_mimeutil.c: comment - out some code that is not used and does not have prototypes - -2007-07-05 22:32 +0000 [r73552] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 73551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73551 | russell | 2007-07-05 17:31:31 -0500 (Thu, 05 Jul 2007) | - 6 lines * Store the call number that a thread is processing - without the full frame bit set to ease debugging * When deferring - a full frame for processing, stick it into the queue for the - thread that is processing frames for that call, not the one that - read the current frame and is about to go back into the idle list - (related to issue #9937) ........ - -2007-07-05 22:29 +0000 [r73550] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73548 | kpfleming | 2007-07-05 17:20:44 -0500 - (Thu, 05 Jul 2007) | 10 lines Merged revisions 73547 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 - Jul 2007) | 2 lines we shouldn't allow G.723.1 endpoints to use - VAD, just like we don't support it for G.729 ........ - ................ - -2007-07-05 22:23 +0000 [r73549] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Add the ability to play an announcement to - queue caller just before bridging Issue 7479, patch by - tristan_mahe. - -2007-07-05 20:52 +0000 [r73513-73514] Russell Bryant <russell@digium.com> - - * main/ast_expr2.y, main/ast_expr2.c: resolve a compiler warning so - i can build in dev mode - - * /, res/res_features.c: Merged revisions 73512 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) | - 5 lines Pass HOLD and UNHOLD frames to the other channel when - they are returned from a native bridge function. This fixes a - problem where when two zap channels are natively bridged and one - does a flash hook, the other channel did not receive music on - hold. (Reported to me directly by Doug Bailey at Digium) ........ - -2007-07-05 19:20 +0000 [r73468] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73467 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, - 05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 - lines Copy language information to the dialog structure when - calling a peer for situations where a PBX may be started on the - dialed channel. (issue #10121 reported by clegall_proformatique) - ........ ................ - -2007-07-05 18:15 +0000 [r73449] Steve Murphy <murf@digium.com> - - * main/pbx.c, utils/expr2.testinput, main/ast_expr2.h, - main/ast_expr2.y, main/ast_expr2f.c, include/asterisk/ast_expr.h, - pbx/pbx_ael.c, UPGRADE.txt, doc/tex/channelvariables.tex, - utils/ael_main.c, main/ast_expr2.fl, main/ast_expr2.c, - utils/check_expr.c: In regards to changes for 9508, expr2 system - choking on floating point numbers, I'm adding this update to - round out (no pun intended) and make this FP-capable version of - the Expr2 stuff interoperate better with previous integer-only - usage, by providing Functions syntax, with 20 builtin functions - for floating pt to integer conversions, and some general floating - point math routines that might commonly be used also. Along with - this, I made it so if a function was not a builtin, it will try - and find it in the ast_custom_function list, and if found, - execute it and collect the results. Thus, you can call system - functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs, - without having to wrap them in $\{...\} (curly brace) notation. - Did a valgrind on the standalone and made sure there's no mem - leaks. Looks good. Updated the docs, too. - -2007-07-05 17:21 +0000 [r73432] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Remove directory creation of directories - we've never used. - -2007-07-05 16:05 +0000 [r73402] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: Merged revisions 73400 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73400 | mmichelson | 2007-07-05 10:59:41 -0500 (Thu, 05 Jul - 2007) | 5 lines Correcting a minor CLI bug I found. When issuing - the queue show command, if you type queue show and then press - tab, you can continue pressing tab and it will keep - auto-completing queue names even though only 1 queue can be used - as an argument. ........ - -2007-07-05 15:29 +0000 [r73399] Russell Bryant <russell@digium.com> - - * channels/chan_vpb.cc, /, channels/Makefile: Merged revisions - 73398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r73398 | russell | 2007-07-05 10:28:27 -0500 (Thu, 05 Jul 2007) | - 2 lines Make this module build for me in dev-mode ........ - -2007-07-05 14:22 +0000 [r73317-73359] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /, apps/app_chanspy.c: Merged revisions 73355 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73355 | file | 2007-07-05 11:21:44 -0300 (Thu, - 05 Jul 2007) | 10 lines Merged revisions 73349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 - lines Tweak spy locking. (issue #9951 reported by welles) - ........ ................ - - * channels/chan_local.c, /: Merged revisions 73319 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73319 | file | 2007-07-05 10:27:40 -0300 (Thu, - 05 Jul 2007) | 10 lines Merged revisions 73318 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 - lines Actually check to make sure a PBX was started on one of the - Local channels instead of blindly assuming it was. (issue #10112 - reported by makoto) ........ ................ - - * /, apps/app_queue.c: Merged revisions 73316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73316 | file | 2007-07-05 10:22:13 -0300 (Thu, - 05 Jul 2007) | 10 lines Merged revisions 73315 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 - lines Reset ServicelevelPerf variable back to 0 if we are unable - to calculate it each time... otherwise we will get previous - values. (issue #10117 reported by noriyuki) ........ - ................ - -2007-07-05 07:45 +0000 [r73209-73298] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, - configs/misdn.conf.sample, channels/misdn_config.c: added general - Jitterbuffer Implementation. #9960 - - * /, channels/misdn/isdn_lib.c: Merged revisions 73253 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73253 | crichter | 2007-07-04 16:53:48 +0200 - (Mi, 04 Jul 2007) | 9 lines Merged revisions 73252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 - Jul 2007) | 1 line bchannel configurations like echocancel and - volume control, need to be setuped on inbound calls too. ........ - ................ - - * channels/chan_misdn.c, /: Merged revisions 73208 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73208 | crichter | 2007-07-04 10:27:44 +0200 - (Mi, 04 Jul 2007) | 9 lines Merged revisions 73207 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 - Jul 2007) | 1 line bad bug in overlapdial case, we called - start_pbx multiple times, because the state wasn't changed.. - ........ ................ - -2007-07-03 22:17 +0000 [r73191] Steve Murphy <murf@digium.com> - - * /: blocking 73143 (revert of 9508 bug fix for 1.4) -- don't want - it backed out of trunk, too - -2007-07-03 21:44 +0000 [r73144-73175] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: mkstemp doesn't specify a file mode, so we - should chmod it to VOICEMAIL_FILE_MODE Taken from a larger patch - by ltd - the rest of which is no longer necessary in trunk. - Closes issue #9231 - - * apps/app_meetme.c: Fix a build warning, and potential issue if - option p is not set at all. - - * apps/app_meetme.c: Add support for changing the exit key from # - to any DTMF. This does not break existing configs - the arguments - to p are optional. Issue 8827, initial patch by junky, mostly - rewritten by fw to re-use option p, further modified by me. - -2007-07-03 18:25 +0000 [r73127] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix up the device state processing thread in - app_queue so that it's not possible for there to be entries in - the queue and the thread is just sleeping (Thanks to mmichelson - for bringing the problem to my attention) - -2007-07-03 12:40 +0000 [r73054] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, /: Merged revisions 73053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500 - (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 - Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it - does not, because atoi does not distinguish between 0 and error - (closes issue #10106) ........ ................ - -2007-07-03 08:22 +0000 [r73006] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 73005 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r73005 | crichter | 2007-07-03 10:17:06 +0200 - (Di, 03 Jul 2007) | 9 lines Merged revisions 73004 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 - Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only - be called from mISDN Source channels.. #9449 ........ - ................ - -2007-07-03 05:21 +0000 [r73003] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Typo (closes issue 10105) - -2007-07-03 02:51 +0000 [r72987] Jason Parker <jparker@digium.com> - - * res/res_jabber.c: Correct an issue where the wrong type was being - used to start sasl. Pointed out by and patch provided by mog. - -2007-07-02 23:02 +0000 [r72982-72986] Russell Bryant <russell@digium.com> - - * main/pbx.c, doc/tex/ast_funcdocs.tex (removed), main/manager.c, - doc/tex/ast_cli_commands.tex (removed), res/res_agi.c, - doc/tex/ast_appdocs.tex (removed), doc/tex/asterisk.tex, - doc/tex/ast_manager_actiondocs.tex (removed), - doc/tex/ast_agi_commands.tex (removed), main/cli.c: After some - discussion on the asterisk-dev list, we determined that this - approach for extracting application, function, manager, and agi - documentation is the wrong one to take. The most severe problem - is that the output depends on which modules are loaded as well as - compile time options, which both determine which parts are - available. - - * doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), - doc/tex/ast_cli_commands.tex (added), doc/tex/ast_appdocs.tex - (added), doc/tex/realtime.tex (added), doc/qos.tex (removed), - doc/queues-with-callback-members.tex (removed), doc/tex/dundi.tex - (added), doc/ajam.tex (removed), doc/tex/cliprompt.tex (added), - doc/misdn.tex (removed), doc/manager.tex (removed), - doc/tex/chaniax.tex (added), doc/sla.tex (removed), - doc/billing.tex (removed), doc/tex/app-sms.tex (added), - build_tools/prep_tarball, doc/tex/ices.tex (added), - doc/localchannel.tex (removed), doc/cdrdriver.tex (removed), - doc/tex/asterisk.tex (added), doc/tex/queuelog.tex (added), - doc/freetds.tex (removed), doc/odbcstorage.tex (removed), - doc/tex/hardware.tex (added), doc/tex/mp3.tex (added), doc/tex - (added), doc/channelvariables.tex (removed), doc/ael.tex - (removed), doc/enum.tex (removed), doc/tex/configuration.tex - (added), doc/security.tex (removed), doc/tex/asterisk-conf.tex - (added), Makefile, doc/imapstorage.tex (removed), - doc/tex/ast_funcdocs.tex (added), doc/privacy.tex (removed), - doc/tex/ast_manager_actiondocs.tex (added), - doc/ast_agi_commands.tex (removed), doc/tex/jitterbuffer.tex - (added), doc/ast_cli_commands.tex (removed), - doc/tex/extensions.tex (added), doc/ast_appdocs.tex (removed), - doc/tex/queues-with-callback-members.tex (added), doc/tex/qos.tex - (added), doc/realtime.tex (removed), doc/dundi.tex (removed), - doc/tex/ajam.tex (added), doc/cliprompt.tex (removed), - doc/tex/manager.tex (added), doc/tex/misdn.tex (added), - doc/chaniax.tex (removed), doc/tex/README.txt (added), - doc/tex/sla.tex (added), doc/app-sms.tex (removed), - doc/tex/billing.tex (added), doc/ices.tex (removed), - doc/tex/localchannel.tex (added), doc/tex/cdrdriver.tex (added), - doc/asterisk.tex (removed), doc/queuelog.tex (removed), - doc/tex/odbcstorage.tex (added), doc/tex/freetds.tex (added), - doc/hardware.tex (removed), doc/mp3.tex (removed), - doc/tex/channelvariables.tex (added), doc/tex/ael.tex (added), - doc/tex/enum.tex (added), doc/configuration.tex (removed), - doc/tex/security.tex (added), doc/asterisk-conf.tex (removed), - doc/tex/imapstorage.tex (added), doc/ast_funcdocs.tex (removed), - doc/tex/privacy.tex (added), doc/tex/Makefile (added), - doc/ast_manager_actiondocs.tex (removed), - doc/tex/ast_agi_commands.tex (added): * Move LaTeX docs into a - tex/ subdirectory of the doc/ dir * Add a Makefile in doc/tex/ - for generating PDF and HTML * Add a README.txt file to doc/tex/ - to document which tools are used and what web sites to visit for - getting them. * Update build_tools/prep_tarball to put the proper - Asterisk version string in the automatically generated PDF for - release tarballs - -2007-07-02 21:50 +0000 [r72940] Steve Murphy <murf@digium.com> - - * utils/expr2.testinput, /, main/Makefile, main/ast_expr2.h, - main/ast_expr2.y, main/ast_expr2f.c, UPGRADE.txt, - main/ast_expr2.fl, main/ast_expr2.c: Merged revisions 72933 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 - line support for floating point numbers added to ast_expr2 - $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp - numbers. The MATH function returns fp numbers by default, so this - fix is considered necessary. ........ - -2007-07-02 20:45 +0000 [r72937-72939] Russell Bryant <russell@digium.com> - - * res/res_agi.c, doc/ast_agi_commands.tex: Fix up the AGI doc dump - CLI command and update the AGI commands tex file to not include a - bunch of empty entries. - - * doc/ast_cli_commands.tex (added), doc/asterisk.tex: Add CLI - commands to the docs - - * main/cli.c: Add a CLI command to output docs on CLI commands to a - file - -2007-07-02 20:35 +0000 [r72935-72936] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Yet another Solaris tweak... - - * res/res_limit.c: Fix building under Solaris. - -2007-07-02 19:31 +0000 [r72920-72932] Russell Bryant <russell@digium.com> - - * doc/asterisk.tex, doc/ast_agi_commands.tex (added): Add AGI - commands to the documentation - - * res/res_agi.c: Add a CLI command to export the AGI command docs - - * res/res_agi.c: Add a note that the AGI commands array is not - handled in a thread-safe way - - * doc/asterisk.tex, doc/ast_manager_actiondocs.tex (added): Update - the documentation to include a manager action reference - - * main/manager.c: Add a CLI command to dump the built-in manager - action documentation - - * main/manager.c, /: Merged revisions 72926 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72926 | russell | 2007-07-02 13:18:46 -0500 (Mon, 02 Jul 2007) | - 3 lines Remove a bogus comment and add proper locking to the - handler function for the CLI command to show information on - manager actions. ........ - - * doc/ast_funcdocs.tex (added), doc/asterisk.tex: update - documentation to include dialplan functions - - * main/pbx.c: Add "core dump funcdocs" CLI command - - * main/pbx.c: change the "core dump appdocs" CLI command to use the - new API for creating CLI commands - - * doc/ast_appdocs.tex: update application documentation dump - -2007-07-02 14:39 +0000 [r72889] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 72888 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2 - lines Added additional DTMF debug messages for when emulation - occurs. ........ - -2007-07-02 09:34 +0000 [r72867-72869] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 72852 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72852 | crichter | 2007-07-02 10:41:08 +0200 - (Mo, 02 Jul 2007) | 9 lines Merged revisions 72585 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 - Jun 2007) | 1 line check if the bchannel stack id is already - used, if so don't use it a second time. Also added a release_chan - lock, so that the same chan_list object cannot be freed twice. - chan_misdn does not crash anymore on heavy load with these - changes. ........ ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - Merged revisions 72851 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72851 | crichter | 2007-07-02 10:27:19 +0200 - (Mo, 02 Jul 2007) | 9 lines Merged revisions 72099 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 - Jun 2007) | 1 line simplified generation for dummy bchannels, - also we mark them as dummies, so they are not used later as - real-bchannels, optimized the RESTART mechanisms, we block a - channel now on cause:44, and send out a RESTART automatically, - then on reception of RESTART_ACKNOWLEDGE we unblock the channel - again. ........ ................ - - * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Merged - revisions 72850 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72850 | crichter | 2007-07-02 10:14:43 +0200 - (Mo, 02 Jul 2007) | 9 lines Merged revisions 72087 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 - Jun 2007) | 1 line simplified channel finding and locking a lot. - removed unnecessary #ifdefed areas. ........ ................ - -2007-07-01 23:53 +0000 [r72807] Russell Bryant <russell@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 72806 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72806 | russell | 2007-07-01 18:52:45 -0500 - (Sun, 01 Jul 2007) | 13 lines Merged revisions 72805 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 - Jul 2007) | 5 lines When appending lines to call files to keep - track of retries, write a leading newline just in case the - original call file did not have a newline at the end. This fix is - in response to a problem I saw reported on the asterisk-users - mailing list. ........ ................ - -2007-06-30 16:53 +0000 [r72767] Russell Bryant <russell@digium.com> - - * /, configure, configure.ac: Merged revisions 72766 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r72766 | russell | 2007-06-30 11:50:40 -0500 (Sat, 30 - Jun 2007) | 3 lines Tweak the configure script so that error - output isn't spewed to the console when searching for GTK2 libs, - and they aren't found. ........ - -2007-06-29 21:37 +0000 [r72741] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Add support - for regcontext and regexten to chan_skinny Issue 9762, patch by - mvanbaak. - -2007-06-29 21:24 +0000 [r72738] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/http.c: Fix my recent change for sending large files via the - http server. This code *must* write the file to the FILE *, and - not the raw fd. Otherwise, it breaks TLS support. Thanks to rizzo - for catching this! - -2007-06-29 21:14 +0000 [r72727] Luigi Rizzo <rizzo@icir.org> - - * main/minimime/Makefile: As the comment in the code says: Use - weaker error checking because we have some automatically - generated files. However just mask out -Werror, because other - warnings below: -Wundef -Wstrict-prototypes - -Wmissing-declarations -Wmissing-prototypes may actually be - important and spot out real bugs. - -2007-06-29 20:56 +0000 [r72701-72706] Russell Bryant <russell@digium.com> - - * /, formats/format_pcm.c: Merged revisions 72705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72705 | russell | 2007-06-29 15:56:18 -0500 (Fri, 29 Jun 2007) | - 1 line give format_pcm a more concise destription ........ - - * include/asterisk/http.h, main/manager.c, configure, - include/asterisk/autoconfig.h.in, configure.ac, main/http.c: - Merge changes from team/russell/http_filetxfer Handle - transferring large files from the built-in http server. - Previously, the code attempted to malloc a block as large as the - file itself. Now it uses the sendfile() system call so that the - file isn't copied into userspace at all if it is available. - Otherwise, it just uses a read/write of small chunks at a time. - -2007-06-29 20:33 +0000 [r72700] Luigi Rizzo <rizzo@icir.org> - - * main/Makefile: Make sure that we properly recurse in - subdirectories to check dependencies for libraries. Because these - targets (e.g. minimime/libmmime.a) are real ones, declaring them - .PHONY would cause them to be rebuilt every time (see e.g. SVN - 64355). As a workaround I am using the following CHECK_SUBDIR - target: CHECK_SUBDIR: # do nothing, just make sure that we - recurse in the subdir/ minimime/libmmime.a: CHECK_SUBDIR @cd - minimime && $(MAKE) libmmime.a which seems to do a better job - than .PHONY (probably because .PHONY forces the rebuild even if - the recursive make does not think it is necessary). If this turns - out to be the correct approach, we can then merge it back into - 1.4 - -2007-06-29 20:02 +0000 [r72670] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Found a grievous logical error in - get_vm_state_by_imapuser. The imapuser being passed in was never - getting compared to imapusers of any of the vm_states in the - vmstates list. I also found some places in the code where I used - my typical brace style and changed it to match the typical - Asterisk brace style. - -2007-06-29 19:09 +0000 [r72666] Luigi Rizzo <rizzo@icir.org> - - * /: 72665 not applicable to trunk - -2007-06-29 04:56 +0000 [r72555-72557] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /: Merged revisions 72556 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72556 | tilghman | 2007-06-28 23:47:11 -0500 (Thu, 28 Jun 2007) - | 2 lines Issue 10055 - Change memory allocation to use the heap - for a command, since the output has the potential to overflow the - stack (as it did here) ........ - -2007-06-28 21:31 +0000 [r72539] Jason Parker <jparker@digium.com> - - * Makefile, configure, configure.ac, makeopts.in: Apparently some - builds of gcc don't have declaration-after-statement. This checks - for it in configure, and only uses it if it's available. If it's - wrong, somebody please yell at me and tell me why. - -2007-06-28 20:52 +0000 [r72524] Dwayne M. Hubbard <dhubbard@digium.com> - - * funcs/func_math.c: Added AND, OR, and XOR bitwise operations to - MATH for issue 9891, thanks jcmoore - -2007-06-28 19:41 +0000 [r72492] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c, res/res_config_odbc.c, - include/asterisk/strings.h: Remove the ill-advised ast_restrdupa - API call and related structures - -2007-06-28 19:35 +0000 [r72490-72491] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Fix building with - -Wdeclaration-after-statement, here too - - * res/res_jabber.c: Fix building with -Wdeclaration-after-statement - -2007-06-28 19:07 +0000 [r72452-72466] Luigi Rizzo <rizzo@icir.org> - - * /: 72462 is not applicable to trunk - - * res/res_features.c, apps/app_sms.c: move variable declarations to - the beginning of a block. Not applicable to previous branches. - - * channels/chan_skinny.c: move variable declarations to the - beginning of the block - - * apps/app_minivm.c: move variable declarations to the beginning of - a block. Not applicable to previous branches - - * /: 72453 was already applied to trunk some time ago - - * Makefile: Add -Wdeclaration-after-statement to AST_DEVMODE to - detect declarations in the middle of a block. Approved by: - Russel, Kevin The fallout will be fixed in separate commits. I am - doing this only on trunk only for the time being, because 1.4 - still requires a bit more polishing to give a clean compile (at - least on FreeBSD). - -2007-06-28 16:35 +0000 [r72437] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Fix bug where point code gets corrupted on - CPG - -2007-06-27 23:30 +0000 [r72384] Brett Bryant <bbryant@digium.com> - - * /, main/asterisk.c: Merged revisions 72383 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72383 | bbryant | 2007-06-27 18:29:14 -0500 - (Wed, 27 Jun 2007) | 11 lines Merged revisions 72373 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 - Jun 2007) | 3 lines Reinstating patch. This actually fixes the - problem, however I was running a development branch without it - and mistakenly thought it wasn't fixed. Fixes issue #10010, and - #9654: 100% CPU usage caused by an asterisk console losing it's - controlling terminal. ........ ................ - -2007-06-27 23:26 +0000 [r72354-72382] Joshua Colp <jcolp@digium.com> - - * /, apps/app_mixmonitor.c: Merged revisions 72381 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, - 27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 - lines Update documentation to clarify variable usage with - MixMonitor. (issue #9494 reported by netoguy) ........ - ................ - - * channels/chan_jingle.c: Silly jingle... - - * channels/chan_sip.c, CHANGES: Add SIPREFERRINGCONTEXT and - SIPREFERREDBYHDR variables when a transfer takes place. (issue - #8378 reported by jcovert) - -2007-06-27 23:04 +0000 [r72337] Brett Bryant <bbryant@digium.com> - - * /, main/asterisk.c: Merged revisions 72335 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72335 | bbryant | 2007-06-27 18:03:01 -0500 - (Wed, 27 Jun 2007) | 10 lines Merged revisions 72333 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 - Jun 2007) | 2 lines Reverted changes for earlier revisions 72259 - to 72261. Issue #9654, #10010 ........ ................ - -2007-06-27 22:58 +0000 [r72330-72332] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_gtalk.c: Merged revisions 72331 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun - 2007) | 2 lines Make payload IDs for iLBC/Speex match to our - list. Since these are dynamic payloads the other side shouldn't - care. (issue #9426 reported by irroot) ........ - - * /, apps/app_queue.c: Merged revisions 72328 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72328 | file | 2007-06-27 18:45:49 -0400 (Wed, - 27 Jun 2007) | 10 lines Merged revisions 72327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 - lines Fix issue where queue log events might be missing. (issue - #7765 reported by mtryfoss) ........ ................ - -2007-06-27 22:47 +0000 [r72329] Mark Michelson <mmichelson@digium.com> - - * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: - Added ability to customize which buttons control forward, - reverse, pause, and stop during message playback. (closes issue - 9474, reported and patched by jaroth with modifications by me) - -2007-06-27 22:27 +0000 [r72325-72326] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix a segfault when trying to tab complete the "core - show uptime" command. Reported in #asterisk-dev on IRC by - jcmoore, fixed by me. - - * main/say.c: Add support for Thai language in say.c Issue 9417, - patch by dome, with some cleanup done by me. - -2007-06-27 21:44 +0000 [r72304] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Let's NOT create a deadlock scenario here - -2007-06-27 21:09 +0000 [r72274] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 72272 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72272 | russell | 2007-06-27 16:08:34 -0500 - (Wed, 27 Jun 2007) | 13 lines Merged revisions 72267 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 - Jun 2007) | 5 lines Fix a minor issue with parsing the priority - number. You could have as much whitespace as you want around a - numeric priority, but you couldn't have any whitespace around a - special priority like "n" or "hint". (issue #10039, reported by - mitheloc, fixed by me) ........ ................ - -2007-06-27 20:47 +0000 [r72261] Brett Bryant <bbryant@digium.com> - - * /, main/asterisk.c: Merged revisions 72260 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72260 | bbryant | 2007-06-27 15:46:12 -0500 - (Wed, 27 Jun 2007) | 12 lines Merged revisions 72259 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 - Jun 2007) | 4 lines Fixes 100% load when controlling terminal - disappears. Issue #9654, #10010 ........ ................ - -2007-06-27 20:26 +0000 [r72233-72258] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 72257 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, - 27 Jun 2007) | 10 lines Merged revisions 72256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 - lines I may possibly get shot for doing this... but... defer CDR - processing until after the channel has been dealt with. This - should eliminate all of the issues with channels going funky - (SIP/PRI) when you are posting CDRs to a database that is either - slow or unavailable and do not want to enable batching. ........ - ................ - - * /: Fix up properties. - - * main/logger.c: Fix -T option. (issue #10073 reported by xylome) - -2007-06-27 19:50 +0000 [r72232] Mark Michelson <mmichelson@digium.com> - - * /, configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: - Adding feature to support the storage and retrieval of voicemail - greetings using IMAP storage. This feature may be turned on by - adding imapgreetings=yes to the general section of voicemail.conf - voicemail.conf.sample has details on the options added. As a - result, IMAP storage now has RETRIEVE and DISPOSE macros defined. - In addition to the IMAP greeting changes, I also have added an - enum for the voicemail folders and so now the code should be - easier to understand and maintain when it comes to this area. - -2007-06-27 19:13 +0000 [r72207] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 72205 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72205 | kpfleming | 2007-06-27 14:13:21 -0500 (Wed, 27 Jun 2007) - | 2 lines use the proper type for storing group number bits so - that if someone specifies 'group=42' it will actually work - instead of being silently ignored ........ - -2007-06-27 18:37 +0000 [r72183] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 72182 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72182 | qwell | 2007-06-27 13:36:56 -0500 (Wed, 27 Jun 2007) | 4 - lines Fix another problem in voicemail with missing symbols. - Issue 10074, patch by kryptolus, extended to include #if 0'd - blocks (just in case) ........ - -2007-06-27 17:34 +0000 [r72149] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 72148 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2 - lines Make the ast_read_noaudio API call behave better under - circumstances where DTMF emulation was happening and a generator - was setup. (issue #10065 reported by stevefeinstein) ........ - -2007-06-27 17:14 +0000 [r72134] Jason Parker <jparker@digium.com> - - * /, channels/chan_gtalk.c: Merged revisions 72125 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun - 2007) | 4 lines Don't modify a variable that we don't want - modified. Make a copy of it instead. Issue 10029, patch by - phsultan with slight modifications by me (to remove needless - casts). Note: chan_jingle in trunk does not appear to have the - same bug. ........ - -2007-06-27 16:38 +0000 [r72113] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: Merged revisions 72112 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) | - 3 lines Only output debug information related to RTCP timestamps - when RTCP debug is turned on (issue #10066, patch by me) ........ - -2007-06-27 08:08 +0000 [r72052] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/isdn_lib.c: Merged revisions 72042 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r72042 | crichter | 2007-06-27 09:58:06 +0200 - (Mi, 27 Jun 2007) | 13 lines Merged revisions 72040-72041 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | - 1 line for inbound TE calls, we setup the bchannel when we get - the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. - removed some #if 0 areas which weren't used anymore. ........ - r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | - 1 line isdn_lib.c didn't compile ........ ................ - -2007-06-27 01:00 +0000 [r71988-72007] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 72006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r72006 | file | 2007-06-26 20:58:35 -0400 (Tue, 26 Jun 2007) | 2 - lines Make unloading of pbx_dundi actually work. ........ - - * channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add rtpdest - option to SIP CHANNEL() dialplan function to return the IP - address and port that RTP (be it audio/video/text) is going to. - -2007-06-26 23:03 +0000 [r71952-71954] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 71953 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71953 | mmichelson | 2007-06-26 18:02:09 -0500 (Tue, 26 Jun - 2007) | 4 lines Removing a pointless line. This variable was - already set earlier and between then and this line, there is no - way that the values on the right side of the assignment could - have changed. ........ - - * apps/app_voicemail.c: The variable msgnum was being overwritten - if IMAP storage was enabled. Put necessary #ifndef's around the - line which would overwrite. - -2007-06-26 20:36 +0000 [r71916] Jason Parker <jparker@digium.com> - - * /, main/rtp.c: Merged revisions 71915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4 - lines Don't dereference a pointer that may be NULL here. Issue - 10017. ........ - -2007-06-26 20:34 +0000 [r71883-71914] Mark Michelson <mmichelson@digium.com> - - * apps/app_record.c: Create directory if it does not exist. (Closes - issue 10061, Reported and patched by eliel) - - * /, apps/app_voicemail.c: Merged revisions 71877 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71877 | mmichelson | 2007-06-26 14:00:05 -0500 (Tue, 26 Jun - 2007) | 11 lines A few changes, the ultimate goal of which is to - keep better track of the number of messages that a mailbox - currently has. A description of the changes: 1. Changed the - "updated" field of the vm_state struct to act more as a binary - semaphore than a counting semaphore, since its current - implementation made the inboxcount function not work properly. - This change falls in line with a change made by UPenn with their - IMAP setup and helps to sync our changes with theirs. 2. - Eliminated some redundant calls to get_vm_state_by_mailbox inside - leave_voicemail 3. Use the play_folder variable to keep track of - the number of old and new messages in a mailbox as the messages - are deleted 4. Added an increment to the number of new messages - that was not there previously in the leave_voicemail function - ........ - -2007-06-26 16:39 +0000 [r71830] Jason Parker <jparker@digium.com> - - * res/res_jabber.c: Simplify some code in res_jabber relating to - SASL support. Issue 9988, patch by phsultan. - -2007-06-26 15:50 +0000 [r71797] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 71796 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71796 | mmichelson | 2007-06-26 10:47:31 -0500 (Tue, 26 Jun - 2007) | 5 lines Fixing bug where the authuser was mistakenly - pulled from the mailbox string instead of the IMAP user. (closes - issue 10054, reported and patched by jaroth) ........ - -2007-06-26 12:30 +0000 [r71752] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 71751 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71751 | tilghman | 2007-06-26 07:27:47 -0500 - (Tue, 26 Jun 2007) | 10 lines Merged revisions 71750 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 - Jun 2007) | 2 lines Issue 10062 - Trying to move a message - without selecting one first results in memory corruption ........ - ................ - -2007-06-26 00:10 +0000 [r71721-71732] Mark Michelson <mmichelson@digium.com> - - * configure, configure.ac: Fixes a problem where Asterisk would not - compile if IMAP_STORAGE was enabled. Needed to add a space - between file name and options. - - * apps/app_voicemail.c: In my commit earlier today, I accidentally - left a prototype that isn't defined. This gets rid of that - prototype. - -2007-06-25 19:20 +0000 [r71688] Russell Bryant <russell@digium.com> - - * doc/imapstorage.tex, configure, configure.ac, - apps/app_voicemail.c: Allow compilation off app_voicemail with - IMAP_STORAE against a system installed version of the c-client - library. (issue #10047, jcollie) - -2007-06-25 18:20 +0000 [r71658] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: Merged revisions 71657 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71657 | tilghman | 2007-06-25 13:14:59 -0500 - (Mon, 25 Jun 2007) | 10 lines Merged revisions 71656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 - Jun 2007) | 2 lines Issue 10035 - handle_exec returns a result - inconsistent with all of the other AGI commands ........ - ................ - -2007-06-25 16:43 +0000 [r71637] Steve Murphy <murf@digium.com> - - * main/cdr.c: Luigi's suggestion to move the llfrom decl was a good - one. Done. - -2007-06-25 16:13 +0000 [r71630] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Using inboxcount instead of countmessages. - -2007-06-25 15:35 +0000 [r71577-71613] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Tweak CLI command completion and some help - text. (issue #10049 reported by IgorG) - - * /, channels/chan_h323.c: Merged revisions 71576 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71576 | file | 2007-06-25 10:13:45 -0400 (Mon, 25 Jun 2007) | 2 - lines Build a peer as well when hash323 is enabled in users.conf - (issue #9599 reported by asagage) ........ - -2007-06-25 13:42 +0000 [r71557] Russell Bryant <russell@digium.com> - - * main/say.c, main/rtp.c, main/sched.c: Convert so more logging to - ast_debug (issue #10045, dimas) - -2007-06-25 13:04 +0000 [r71521-71525] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 71522 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r71522 | file | 2007-06-25 09:03:03 -0400 (Mon, 25 Jun - 2007) | 2 lines Minor tweak for queueing up the unhold frame... - this will teach me to do bugs while half asleep. (issue #10046 - reported by dimas) ........ - - * res/res_agi.c: Minor header inclusion tweak for new usage of - stat() - -2007-06-25 12:40 +0000 [r71520] Russell Bryant <russell@digium.com> - - * doc/asterisk-mib.txt, /: Merged revisions 71519 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71519 | russell | 2007-06-25 07:40:06 -0500 (Mon, 25 Jun 2007) | - 2 lines Fix a typo in the Asterisk mib. (issue #10048, Matti) - ........ - -2007-06-25 09:46 +0000 [r71475-71500] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_lib.c: Merged revisions 71214 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71214 | crichter | 2007-06-23 00:44:42 +0200 - (Sa, 23 Jun 2007) | 9 lines Merged revisions 70341 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 - Jun 2007) | 1 line fixed a bug that was introduced by copy and - paste in the last commit ..bchannels weren't cleaned properly. - ........ ................ - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 71123 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71123 | crichter | 2007-06-22 17:38:08 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 70672 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 - Jun 2007) | 1 line we activate the bchannels in TE mode on - incoming calls only when we want to connect the call. ........ - ................ - - * /, channels/misdn/isdn_lib.c: Merged revisions 71122 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71122 | crichter | 2007-06-22 17:34:31 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 70342 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 - Jun 2007) | 1 line forgot one place .. ........ ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_lib.c: Merged revisions 71121 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71121 | crichter | 2007-06-22 17:32:54 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 70311 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 - Jun 2007) | 1 line on receiption of cause:44 we mark the channel - as in use and inform the user about the situation, we need to - test the RESTART stuff then. Also shuffled the - empty_chan_in_stack function after the bchannel cleaning - functions, to avoid race conditions. ........ ................ - - * channels/chan_misdn.c, /: Merged revisions 71120 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71120 | crichter | 2007-06-22 17:30:08 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 69887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 - Jun 2007) | 1 line when we send out a SETUP, but get no response, - we should cleanup everything after reception of a hangup. - ........ ................ - - * /, channels/misdn/isdn_msg_parser.c: Merged revisions 71118 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71118 | crichter | 2007-06-22 17:27:53 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 69053 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 - Jun 2007) | 1 line restart indicator 0x80 is correct, at least - that's what libpri does. ........ ................ - - * channels/chan_misdn.c, /: Merged revisions 71106 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71106 | crichter | 2007-06-22 17:22:06 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 68887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 - Jun 2007) | 1 line if the bridged partner is mISDN too we should - not send dtmf tones, they are transmitted inband always ........ - ................ - - * channels/chan_misdn.c, /: Merged revisions 71096 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71096 | crichter | 2007-06-22 17:17:04 +0200 - (Fr, 22 Jun 2007) | 9 lines Merged revisions 68874 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 - Jun 2007) | 1 line if we have already some digits, we just stop - the tones. ........ ................ - -2007-06-25 01:11 +0000 [r71413-71434] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 71430 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, - 24 Jun 2007) | 10 lines Merged revisions 71414 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 - lines Ignore other URIs after the first in a 300 Multiple Choice - response. (issue #10041 reported by homesick) ........ - ................ - - * main/cdr.c, /: Merged revisions 71422 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71422 | file | 2007-06-24 21:07:31 -0400 (Sun, 24 Jun 2007) | 2 - lines Fix it so 1.4 actually compiles on my box. ........ - - * /, channels/chan_agent.c: Merged revisions 71412 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r71412 | file | 2007-06-24 20:49:21 -0400 (Sun, 24 Jun - 2007) | 2 lines Check to make sure the channel pointer is present - before queueing up an unhold frame on it. (issue #10046 reported - by dimas) ........ - -2007-06-24 20:17 +0000 [r71338-71372] Russell Bryant <russell@digium.com> - - * /, build_tools/prep_tarball: Merged revisions 71371 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r71371 | russell | 2007-06-24 15:16:32 -0500 (Sun, 24 - Jun 2007) | 3 lines Include the menuselect-tree file in tarballs - to make builds from tarballs a little bit faster ........ - - * /, main/asterisk.c: Merged revisions 71362 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71362 | russell | 2007-06-24 15:06:31 -0500 - (Sun, 24 Jun 2007) | 10 lines Merged revisions 71358 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 - Jun 2007) | 2 lines Revert the patch from issue 9654 due to an - unexpected side effect ........ ................ - - * main/udptl.c, apps/app_meetme.c, main/say.c, main/translate.c, - main/jitterbuf.c, apps/app_test.c, main/rtp.c, main/loader.c, - main/io.c, main/manager.c, apps/app_skel.c, apps/app_minivm.c, - main/logger.c, main/http.c, apps/app_rpt.c, main/sched.c: - Conversions to ast_debug() (issue #9984, patches from eliel and - dimas) - -2007-06-24 17:51 +0000 [r71268-71292] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_features.c: Merged revisions 71291 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71291 | tilghman | 2007-06-24 12:50:24 -0500 (Sun, 24 Jun 2007) - | 2 lines Issue 10044 - chan->cdr is NULL here, so peer->cdr is - what we really wanted to use ........ - - * main/manager.c, /, main/db.c: Merged revisions 71289 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71289 | tilghman | 2007-06-24 12:39:34 -0500 - (Sun, 24 Jun 2007) | 10 lines Merged revisions 71288 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 - Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to - be able to set variables to the empty string. ........ - ................ - - * apps/app_mixmonitor.c: Issue 9970 - Ensure directory exists - before trying to write an output file - -2007-06-23 03:32 +0000 [r71231] Steve Murphy <murf@digium.com> - - * main/cdr.c, /, res/res_features.c: Merged revisions 71230 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71230 | murf | 2007-06-22 21:29:48 -0600 (Fri, 22 Jun 2007) | 1 - line This patch is meant to fix 8433; where clid and src are lost - via bridging. ........ - -2007-06-22 19:53 +0000 [r71190] Tilghman Lesher <tlesher@digium.com> - - * apps/app_sms.c: Code cleanups - -2007-06-22 16:19 +0000 [r71146-71158] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Use stat to determine whether the file exists or - not. (issue #10038 reported by Mike Anikienko) - - * main/rtp.c: Behold the magic of casting! - -2007-06-22 15:15 +0000 [r71093] Steve Murphy <murf@digium.com> - - * main/cdr.c, /, main/rtp.c: Merged revisions 71063 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun - 2007) | 1 line My conditions for merging amaflags info was naive; - DOCUMENTATION is the default, although null is possible; theft of - user-settable fields is not good. Just copy them, leave them - alone. This is for bug 10016. (plus a small fix to rtp, to elim a - compiler warning (dev mode)) ........ - -2007-06-22 15:03 +0000 [r71069] Jason Parker <jparker@digium.com> - - * /, res/res_agi.c, main/file.c, apps/app_speech_utils.c: Merged - revisions 71068 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, - 22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 - lines Fix a few silly usages of ast_playstream() - it only ever - returns 0... Issue 10035 ........ ................ - -2007-06-22 14:56 +0000 [r71067] Brett Bryant <bbryant@digium.com> - - * /, main/asterisk.c: Merged revisions 71066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r71066 | bbryant | 2007-06-22 09:53:08 -0500 - (Fri, 22 Jun 2007) | 18 lines Merged revisions 71064 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 - Jun 2007) | 10 lines Fixed infinite loop when controlling - terminal was lost and return value of input function wasn't - checked for errors. This would cause 100% cpu to be taken up. - (closes issue #9654, issue #10010) Reported by: mnicholson, and - eserra Idea for the patch from mnicholson, patched by me ........ - ................ - -2007-06-22 04:35 +0000 [r71040] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, include/asterisk/utils.h, pbx/pbx_spool.c, - apps/app_dictate.c, apps/app_minivm.c, apps/app_test.c, - main/logger.c, main/utils.c, apps/app_sms.c, res/res_monitor.c, - apps/app_voicemail.c: Issue 9990 - New API ast_mkdir, which - creates parent directories as necessary (and is faster than an - outcall to mkdir -p) - -2007-06-22 04:13 +0000 [r71024] Jason Parker <jparker@digium.com> - - * build_tools/cflags.xml, main/asterisk.c: Nothing to see here. - -2007-06-22 03:15 +0000 [r71004] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 71003 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r71003 | russell | 2007-06-21 22:14:41 -0500 (Thu, 21 Jun 2007) | - 3 lines Fix a small typo which ... well ... completely broke - chan_iax2. oops! (issue #9937, patch by me) ........ - -2007-06-21 23:07 +0000 [r70961] Jason Parker <jparker@digium.com> - - * main/manager.c, configs/manager.conf.sample, - include/asterisk/manager.h, main/rtp.c: Add manager events for - RTCP statistics. Also adds a new "reporting" permission for - manager, since it can be incredibly spammy. This permission was - discussed on the -dev mailing list some months back. Issue 8613, - patch by johann8384, with some minor changes by me. - -2007-06-21 22:41 +0000 [r70951] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 70949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70949 | murf | 2007-06-21 16:34:41 -0600 (Thu, - 21 Jun 2007) | 9 lines Merged revisions 70948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 - line This little fix is in response to bug 10016, but may not - cure it. The code is wrong, clearly. In a situation where you set - the CDR's amaflags, and then ForkCDR, and then set the new CDR's - amaflags to some other value, you will see that all CDRs have had - their amaflags changed. This is not good. So I fixed it. ........ - ................ - -2007-06-21 21:41 +0000 [r70900] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 70899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70899 | file | 2007-06-21 17:40:19 -0400 (Thu, - 21 Jun 2007) | 10 lines Merged revisions 70898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 - lines Don't explode if the gain option is specified without a - value. (issue #9274 reported by mfarver) ........ - ................ - -2007-06-21 21:16 +0000 [r70877-70887] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 70883 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70883 | russell | 2007-06-21 16:14:53 -0500 (Thu, 21 Jun 2007) | - 3 lines Put the thread reading from the socket back in the idle - list if it deferred the processing of a full frame to another - thread ........ - - * /, channels/chan_iax2.c: Merged revisions 70866 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70866 | russell | 2007-06-21 16:07:04 -0500 (Thu, 21 Jun 2007) | - 5 lines If a full frame is received while one of the iax2 threads - is in the middle of handling a full frame for the same call, - queue it up for processing by that same thread later instead of - dropping it. (issue #9937, patch by me) ........ - -2007-06-21 20:28 +0000 [r70857] Steve Murphy <murf@digium.com> - - * /, cdr/cdr_custom.c: Merged revisions 70841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70841 | murf | 2007-06-21 14:19:36 -0600 (Thu, - 21 Jun 2007) | 9 lines Merged revisions 70804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 - line it was pointed out that the cdr_custom config load could get - a lock, and under certain circumstances, would never release it. - I also noted that the situation where more than one mapping spec - was warned about, but did not ignore further mappings as it had - promised. I think I have fixed both situations. ........ - ................ - -2007-06-21 19:54 +0000 [r70809] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 70808 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70808 | mmichelson | 2007-06-21 14:49:44 -0500 (Thu, 21 Jun - 2007) | 4 lines When volgain is used don't leave a temporary file - behind. (Closes Issue 8514, Reported and patched by ulogic, code - reviewed by Jason Parker) ........ - -2007-06-21 19:08 +0000 [r70794] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_buildopts_h: when we are building modules that - other modules depend on, create preprocessor defines (in - buildopts.h) marking that those modules were built - -2007-06-21 18:40 +0000 [r70783] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Merge changes from team/russell/sla_reload * - Add support for the reload of sla.conf (closes issue #9481, patch - by me) - -2007-06-21 18:03 +0000 [r70769] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Remove deprecated function call - -2007-06-21 15:58 +0000 [r70729-70731] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Expand AGISTATUS variable to include NOTFOUND - which is set when the AGI file could not be found. (issue #9285 - reported by srdjan) - - * /, main/rtp.c: Merged revisions 70727 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 - lines Do not Packet2Packet bridge if packetization settings do - not allow it. (issue #9117 reported by phsultan) ........ - -2007-06-21 15:23 +0000 [r70728] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 70726 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) | - 2 lines Remove a couple of duplicate unlocks ........ - -2007-06-21 14:00 +0000 [r70678] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 70677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70677 | file | 2007-06-21 09:58:36 -0400 (Thu, 21 Jun 2007) | 2 - lines Fix building with ODBC storage enabled. (issue #10025 - reported by denisgalvao) ........ - -2007-06-21 13:18 +0000 [r70676] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 70656 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70656 | murf | 2007-06-21 07:00:39 -0600 (Thu, 21 Jun 2007) | 1 - line Via complaints aired in asterisk-users, I submit these - changes, which allow cdr updates to see macro context/exten, - whether hung up or not ........ - -2007-06-20 23:33 +0000 [r70613] Jason Parker <jparker@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 70612 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70612 | qwell | 2007-06-20 18:32:39 -0500 (Wed, 20 Jun 2007) | 4 - lines Fix some potential memory leaks in cdr_pgsql. Issue 10020, - patch by me, with credit to prashant_jois for pointing out the - problem. ........ - -2007-06-20 23:31 +0000 [r70611] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removed an extraneous debug message I'd - left in my previous commit - -2007-06-20 23:31 +0000 [r70610] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, apps/app_queue.c: Fix trunk brokenness; also, - optimize application registration - -2007-06-20 23:26 +0000 [r70607] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/pbx.c, apps/app_queue.c: Cleaning up a - small disaster I created earlier - -2007-06-20 22:55 +0000 [r70555-70561] Jason Parker <jparker@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 70560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70560 | qwell | 2007-06-20 17:55:21 -0500 (Wed, 20 Jun 2007) | 1 - line Fix a stupid mistake in my last cdr_pgsql race condition fix - ........ - - * /, cdr/cdr_pgsql.c: Merged revisions 70554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70554 | qwell | 2007-06-20 17:31:35 -0500 (Wed, 20 Jun 2007) | 4 - lines Fix a race condition in cdr_pgsql that can occur when - reloading the module. Issue 10022, patch by me, with credit to - prashant_jois for finding the bug. ........ - -2007-06-20 22:24 +0000 [r70553] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 70552 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, - 20 Jun 2007) | 10 lines Merged revisions 70551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 - lines Don't overwrite the configured username setting upon a - REGISTER. (issue #8565 reported by jsmith) ........ - ................ - -2007-06-20 21:38 +0000 [r70531] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, apps/app_queue.c: As per 9228, now app_queue - should have the proper machinery to do gosubs. - -2007-06-20 21:31 +0000 [r70530] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Main fix: Fixing a bug which caused - VoiceMailMain to always report that you had 0 messages when using - IMAP storage. Secondary fixes: adding locks to list access in - several places Big thanks to Russell Bryant for helping out with - this. - -2007-06-20 20:54 +0000 [r70493-70495] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 70494 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r70494 | qwell | 2007-06-20 15:53:16 -0500 (Wed, 20 Jun - 2007) | 4 lines Make sure we clear the previously dialed number - if it did not exist. Issue 9958. ........ - - * main/http.c: Revert the change made in revision 45474, since this - causes other issues. Issue 10021. - -2007-06-20 20:10 +0000 [r70461] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael_lex.c, - pbx/pbx_ael.c, doc/ael.tex, include/asterisk/ael_structs.h, - pbx/ael/ael.tab.h, CHANGES, pbx/ael/ael.flex: This finishes the - changes for making Macro args LOCAL to the call, and allowing - users to declare local variables. - -2007-06-20 19:30 +0000 [r70446] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, /: Merged revisions 70445 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70445 | tilghman | 2007-06-20 14:29:23 -0500 - (Wed, 20 Jun 2007) | 10 lines Merged revisions 70444 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 - Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong - channel ........ ................ - -2007-06-20 18:48 +0000 [r70398] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 70397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70397 | russell | 2007-06-20 13:46:49 -0500 - (Wed, 20 Jun 2007) | 13 lines Merged revisions 70396 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 - Jun 2007) | 5 lines Fix a problem where an established call would - not be properly disconnected when a PRI disconnect is received - depending on which cause code was received. (issue #9588, - original patch by softins, updated patch from jtexter3, and some - additional feedback from mhardeman) ........ ................ - -2007-06-20 17:55 +0000 [r70361] Joshua Colp <jcolp@digium.com> - - * main/frame.c, /, main/rtp.c: Merged revisions 70360 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun - 2007) | 2 lines Put the speex packetization values back in but - disable it when setting up the smoother. ........ - -2007-06-20 17:35 +0000 [r70358] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, pbx/pbx_ael.c: Merge work to make U(...) option - work for Dial - -2007-06-20 14:33 +0000 [r70310] Olle Johansson <oej@edvina.net> - - * channels/chan_zap.c: Show TDD status in "zap show channels" - Inspired by work at Omnitor in Sweden - -2007-06-20 13:00 +0000 [r70253-70291] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Oops, shouldn't have taken that last shortcut - (also add some checks) - - * apps/app_stack.c: Another method of doing local variables, - hopefully a little closer to what codefreeze had in mind - - * apps/app_stack.c: Local variables for codefreeze - -2007-06-20 02:13 +0000 [r70234] Russell Bryant <russell@digium.com> - - * /, contrib/scripts/ast_grab_core: Merged revisions 70164 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70164 | russell | 2007-06-19 19:03:22 -0500 (Tue, 19 Jun 2007) | - 2 lines don't delete the backtrace in ast_grab_core ........ - -2007-06-20 00:26 +0000 [r70199] Joshua Colp <jcolp@digium.com> - - * main/frame.c, /: Merged revisions 70198 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70198 | file | 2007-06-19 20:24:36 -0400 (Tue, 19 Jun 2007) | 2 - lines Don't do packetization/smoother stuff with speex, it - doesn't work. ........ - -2007-06-19 23:38 +0000 [r70122-70162] Steve Murphy <murf@digium.com> - - * CHANGES: Added a little verbage to CHANGES - - * apps/app_dial.c, apps/app_queue.c, apps/app_rpt.c: Via bug9228, - no way to create macros via AEL, and some of the apps allow you - to call macros..., I modded the apps that allow macro calls to - allow gosubs calls also, to make them AEL compliant. - - * UPGRADE.txt, CHANGES: Moved those comments from UPGRADE.txt to - CHANGES. Ooops. - - * UPGRADE.txt: Some UPGRADE.txt comments to cover some enhancements - added today. - - * configs/cdr_manager.conf.sample, cdr/cdr_manager.c: This - enhancement provided via bug 9993, a patch to upgrade cdr_manager - to have cdr_custom capabilities. Many thanks to eserra for this - contribution - -2007-06-19 19:15 +0000 [r70088] Russell Bryant <russell@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 70084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | - 3 lines Only attempt to queue a hangup on the owner channel if it - actually exists. (issue #9795, patch from zandbelt) ........ - -2007-06-19 18:31 +0000 [r70063] Steve Murphy <murf@digium.com> - - * main/channel.c, /: Merged revisions 70062 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, - 19 Jun 2007) | 9 lines Merged revisions 70053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 - line This fixes 9246, where channel variables are not available - in the 'h' exten, on a 'ZOMBIE' channel. The fix is to - consolidate the channel variables during a masquerade, and then - copy the merged variables back onto the clone, so the zombie has - the same vars that the 'original' has. ........ ................ - -2007-06-19 17:09 +0000 [r70006] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 70003 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, - 19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 - lines Handle the CC field in the RTP header. (issue #9384 - reported by DoodleHu) ........ ................ - -2007-06-19 17:07 +0000 [r70001] Steve Murphy <murf@digium.com> - - * include/asterisk/callerid.h, channels/chan_zap.c, - doc/India-CID.txt (added), configs/zapata.conf.sample: These - changes were submitted via bug 6683, to allow CID detection in - India, with carriers that do Polarity/DTMF CID signalling. - -2007-06-19 16:25 +0000 [r69988] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 69987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69987 | file | 2007-06-19 12:24:31 -0400 (Tue, - 19 Jun 2007) | 10 lines Merged revisions 69986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 - lines Update BRIDGEPEER variable if set to the new channel name - when a masquerade happens. (issue #9699 reported by dimas) - ........ ................ - -2007-06-19 15:27 +0000 [r69945] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69944 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | - 10 lines Fix a crash that could occur when handing device state - changes. When the state of a device changes, the device state - thread tells the extension state handling code that it changed. - Then, the extension state code calls the callback in chan_sip so - that it can update subscriptions to that extension. A pointer to - a sip_pvt structure is passed to this function as the call which - needs a NOTIFY sent. However, there was no locking done to ensure - that the pvt struct didn't disappear during this process. (issue - #9946, reported by tdonahue, patch by me, patch updated to trunk - to use the sip_pvt lock wrappers by eliel) ........ - -2007-06-19 15:14 +0000 [r69943] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: Add support for - setting nature of address, presentation, and other related SS7 - number options (#10000) - -2007-06-19 13:56 +0000 [r69850-69896] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 69895 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69895 | file | 2007-06-19 09:55:25 -0400 (Tue, - 19 Jun 2007) | 10 lines Merged revisions 69894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 - lines Perform an extra hangup check just in case. (issue #9589 - reported by bcnit) ........ ................ - - * /, res/res_features.c: Merged revisions 69847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69847 | file | 2007-06-19 09:00:57 -0400 (Tue, - 19 Jun 2007) | 10 lines Merged revisions 69846 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 - lines Add parked call extension AFTER the parking slot has been - announced, otherwise two threads will try to handle the same - channel and it will go kaboom. (issue #9191 reported by japple) - ........ ................ - -2007-06-18 23:28 +0000 [r69808-69809] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Undoing my last commit. I misread the code - before. - - * apps/app_voicemail.c: Cleaned up a section where there were two - consecutive identical if statements. Combined the bodies of the - two into one if. I blame svn merging for this. - -2007-06-18 22:23 +0000 [r69807] Brett Bryant <bbryant@digium.com> - - * apps/app_queue.c: Fixed issue where 'stop gracfeully' was hanging - ... - -2007-06-18 21:58 +0000 [r69806] Joshua Colp <jcolp@digium.com> - - * /, main/callerid.c: Merged revisions 69805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69805 | file | 2007-06-18 17:57:10 -0400 (Mon, 18 Jun 2007) | 2 - lines Fix for building on PowerPC under Linux. ........ - -2007-06-18 19:52 +0000 [r69797] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69796 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) - | 2 lines Issue 10005 - Segfault with missing arguments, plus fix - a missing define for SIP INFO channels ........ - -2007-06-18 19:02 +0000 [r69779-69795] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69794 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 - lines Don't count RTP timeout when involved in a T38 fax session. - (issue #9222 reported by ivoc) ........ - - * /, channels/chan_sip.c: Merged revisions 69775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, - 18 Jun 2007) | 10 lines Merged revisions 69765 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 - lines Set the peer name on the dialog to the one configured in - sip.conf and NOT the username to be used for authentication - attempts. (issue #9967 reported by achauvin) ........ - ................ - -2007-06-18 17:50 +0000 [r69745-69746] Tilghman Lesher <tlesher@digium.com> - - * /, contrib/scripts/safe_asterisk: Merged revisions 69744 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69744 | tilghman | 2007-06-18 12:46:40 -0500 - (Mon, 18 Jun 2007) | 10 lines Merged revisions 69743 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 - Jun 2007) | 2 lines Issue 9998 - Remove SIG prefix, since it's - not supported by ksh ........ ................ - - * apps/app_rpt.c: Janitor for ast_localtime - -2007-06-18 16:56 +0000 [r69705-69709] Joshua Colp <jcolp@digium.com> - - * main/dnsmgr.c, /: Merged revisions 69708 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69708 | file | 2007-06-18 12:51:36 -0400 (Mon, 18 Jun 2007) | 2 - lines Remember the DNS lookup done when dnsmgr is called for the - first time so that it does not needlessly spit out changed - messages when the host really didn't change. ........ - - * main/cdr.c, main/dnsmgr.c, main/asterisk.c: Few more rwlist - conversions... why not. - -2007-06-18 16:35 +0000 [r69691-69703] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /, build_tools/menuselect-deps.in, - configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in, - configure.ac, cdr/cdr_odbc.c, res/res_odbc.c, - apps/app_voicemail.c: Merged revisions 69702 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | - 6 lines To prevent 92138749238754 more reports of "I have - unixodbc installed, but still can't build *_odbc.so!", check for - ltdl directly, instead of just listing it as another library to - include in the unixodbc check in the configure script. This also - makes ltdl show up as a dependency in menuselect so people know - what to go install. (related to issue #9989, patch by me) - ........ - - * /, build_tools/prep_moduledeps: Merged revisions 69689 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69689 | russell | 2007-06-18 11:15:12 -0500 (Mon, 18 Jun 2007) | - 5 lines Change the use of "echo -e" to "printf". On systems where - /bin/sh is not bash, most of the lines in menuselect-tree were - getting a "-e" at the beginning of every line. I'm surprised - nobody noticed this, but I think the XML parser was being very - nice and ignoring them. ........ - -2007-06-18 16:06 +0000 [r69663-69672] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69668 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 - lines Don't defer the BYE till later on a transfer when the - transfer itself goes kaboom and has no hope of working. ........ - - * /, channels/chan_sip.c: Merged revisions 69661 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 - lines Few minor transfer tweaks. We can't unlock something we - never locked, and better handle a specific scenario with doing an - attended transfer between two non-bridged calls. ........ - -2007-06-18 15:46 +0000 [r69662] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 69660 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69660 | russell | 2007-06-18 10:46:14 -0500 (Mon, 18 Jun 2007) | - 2 lines Tweak paths for BSD systems (issue #10001, stuarth) - ........ - -2007-06-18 13:57 +0000 [r69626] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69625 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 - lines Fix issue where it would be possible for the negotiated - codecs to get set back to nothing. (issue #9992 reported by - yehavi) ........ - -2007-06-15 20:21 +0000 [r69583] Russell Bryant <russell@digium.com> - - * /: This was only an issue in 1.4. This issue was fixed in trunk - as a part of bbryant's patch to support named dynamic feature - groups. Merged revisions 69579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69579 | russell | 2007-06-15 15:18:58 -0500 (Fri, 15 Jun 2007) | - 5 lines Fix a silly deadlock in res_features that I found while - debugging on one of blitzrage's test machines. It was one of the - situations where he was seeing hung channels, and may be the - cause of some of the reports from other people. (related to issue - #9235) ........ - -2007-06-15 19:25 +0000 [r69559] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 69558 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri, - 15 Jun 2007) | 2 lines Add support for setting the maximum length - of acceptable DTMF in SpeechBackground. - -2007-06-15 15:36 +0000 [r69519] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 69518 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) | - 5 lines The SLATRUNK_STATUS variable indicated "SUCCESS" for both - an answer of the incoming call on the trunk, or if the trunk - reached its ring timeout. This patch changes the variable to say - "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle, - patch by me) ........ - -2007-06-14 23:23 +0000 [r69471] Jason Parker <jparker@digium.com> - - * /, main/config.c: Merged revisions 69470 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69470 | qwell | 2007-06-14 18:22:51 -0500 (Thu, - 14 Jun 2007) | 12 lines Merged revisions 69469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 - lines Fix an issue where the line number in an unterminated - comment block error message would show the wrong line number. - "Reported" to me on #asterisk (somebody posted an error message, - and I happened to catch it) ........ ................ - -2007-06-14 23:01 +0000 [r69436] Russell Bryant <russell@digium.com> - - * main/pbx.c, channels/chan_vpb.cc, apps/app_meetme.c, - res/res_features.c, channels/iax2-provision.c, main/enum.c, - res/res_monitor.c, apps/app_speech_utils.c, main/loader.c, - main/cli.c, main/channel.c, channels/chan_misdn.c, - apps/app_minivm.c, main/http.c, main/file.c, - channels/chan_h323.c, res/res_indications.c, - apps/app_directory.c, main/asterisk.c: Convert uses of strdup() - to ast_strdup() (issue #9983, eliel) - -2007-06-14 22:56 +0000 [r69435] Jason Parker <jparker@digium.com> - - * /, sounds/Makefile: Merged revisions 69434 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69434 | qwell | 2007-06-14 17:56:09 -0500 (Thu, 14 Jun 2007) | 1 - line Update to latest versions of sound files. ........ - -2007-06-14 22:09 +0000 [r69394-69405] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/utils.h, main/pbx.c, /, main/say.c, - cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c, - cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, - main/manager.c, cdr/cdr_sqlite.c, apps/app_minivm.c, - main/callerid.c, main/logger.c, main/stdtime/localtime.c, - cdr/cdr_odbc.c, main/asterisk.c, cdr/cdr_manager.c, - channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions - 69392 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) - | 2 lines use ast_localtime() in every place localtime_r() was - being used ........ - - * formats/format_ogg_vorbis.c: oops... somebody patched this module - without compile-testing it... bad :-) - -2007-06-14 21:09 +0000 [r69327-69360] Russell Bryant <russell@digium.com> - - * /, main/say.c: Merged revisions 69358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69358 | russell | 2007-06-14 16:08:23 -0500 (Thu, 14 Jun 2007) | - 3 lines Fix some problems with saying dates and times for the - "tw" langauge (issue #9964, ljmid) ........ - - * CHANGES: update CHANGES for tw support in voicemail - - * apps/app_voicemail.c: Add support for the tw language in - voicemail (issue #9964, ljmid) - - * funcs/func_rand.c, main/frame.c, channels/chan_local.c, - res/res_features.c, apps/app_record.c, funcs/func_strings.c, - apps/app_test.c, main/devicestate.c, apps/app_alarmreceiver.c, - apps/app_ices.c, channels/chan_iax2.c, main/config.c, - res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c, - apps/app_zapras.c, apps/app_amd.c, channels/chan_alsa.c, - cdr/cdr_odbc.c, main/db.c, apps/app_dial.c, formats/format_wav.c, - channels/chan_agent.c, apps/app_disa.c, - formats/format_ogg_vorbis.c, channels/iax2-provision.c, - apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c, - apps/app_zapbarge.c, channels/chan_misdn.c, - channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, - formats/format_g726.c, apps/app_chanspy.c, main/asterisk.c, - apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c, - res/res_musiconhold.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, - apps/app_followme.c, codecs/codec_zap.c, cdr/cdr_radius.c, - res/res_jabber.c, res/res_config_sqlite.c, main/enum.c, - cdr/cdr_csv.c, main/cdr.c, main/channel.c, main/dial.c, - channels/chan_phone.c, apps/app_osplookup.c, apps/app_minivm.c, - res/res_agi.c, apps/app_mp3.c, main/app.c, apps/app_rpt.c, - main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c, - res/res_config_pgsql.c, funcs/func_version.c, - channels/chan_zap.c, funcs/func_db.c, channels/chan_sip.c, - apps/app_festival.c, apps/app_waitforsilence.c, res/res_crypto.c, - res/res_adsi.c, main/acl.c, apps/app_queue.c, cdr/cdr_tds.c, - channels/chan_jingle.c, apps/app_channelredirect.c, - apps/app_directed_pickup.c, main/adsistub.c, main/callerid.c, - main/file.c, channels/chan_h323.c, channels/chan_nbs.c, - apps/app_stack.c, main/dsp.c: Add a massive set of changes for - converting to use the ast_debug() macro. (issue #9957, patches - from mvanbaak, caio1982, critch, and dimas) - -2007-06-14 16:41 +0000 [r69308] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Clean up debug messages a little bit for ss7 - linkset debugging - -2007-06-14 15:43 +0000 [r69261] Brett Bryant <bbryant@digium.com> - - * main/manager.c: Couple of manager ssl options weren't loading - because of a typo. - -2007-06-14 15:25 +0000 [r69260] Jason Parker <jparker@digium.com> - - * funcs/func_groupcount.c, /: Merged revisions 69259 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69259 | qwell | 2007-06-14 10:21:29 -0500 (Thu, - 14 Jun 2007) | 12 lines Merged revisions 69258 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4 - lines Change a quite broken while loop to a for loop, so - "continue;" works as expected instead of eating 99% CPU... Issue - 9966, patch by me. ........ ................ - -2007-06-13 21:20 +0000 [r69223] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 69221 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69221 | file | 2007-06-13 17:17:28 -0400 (Wed, 13 Jun 2007) | 2 - lines Let's make chan_iax2 media only native transfers actually - work. (issue #9376 reported by simone cittadini) ........ - -2007-06-13 20:03 +0000 [r69187] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69183 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | - 9 lines Move the logic for destroying a call when no response is - received to a BYE outside of the block that checks for FLAG_FATAL - to be set. This flag is only set when the packet is transmitted - with the reliability set to XMIT_CRITICAL when the original - packet is transmitted. A BYE is always sent with it set to - XMIT_RELIABLE, meaning this code could never be encountered. This - resulted in seeing some SIP channels that would never go away - with the last packet sent being a BYE. (part of issue #9235, - patch from jcmoore) ........ - -2007-06-13 20:00 +0000 [r69185] Joshua Colp <jcolp@digium.com> - - * /, channels/iax2-parser.c: Merged revisions 69184 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r69184 | file | 2007-06-13 15:58:59 -0400 (Wed, 13 Jun - 2007) | 2 lines Add TXMEDIA to list so that it is properly - displayed during iax2 packet output. ........ - -2007-06-13 19:47 +0000 [r69182] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 69181 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69181 | mmichelson | 2007-06-13 14:41:13 -0500 (Wed, 13 Jun - 2007) | 5 lines Contains a patch for fixing an encoding problem - when using Outlook to view voicemail emails and attachments. This - fix has also been tested on Thunderbird, Evolution, Pine, and - Mutt. (Issue 9336, reported by marwick, patched by mutterc) - ........ - -2007-06-13 19:10 +0000 [r69147] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 69144 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2 - lines Really ignore NULL frames and check whether the channel - hungup or not. (issue #9912 reported by junky) ........ - -2007-06-13 19:05 +0000 [r69137] Jason Parker <jparker@digium.com> - - * channels/chan_agent.c: Completely remove callback mode and all - references to it from chan_agent. Issue 9969, patch by eliel. - -2007-06-13 18:23 +0000 [r69129-69130] Joshua Colp <jcolp@digium.com> - - * include/asterisk/app.h, funcs/func_groupcount.c, main/app.c, - main/cli.c: Use read/write lock based lists for group counting. - - * /, main/app.c: Merged revisions 69128 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r69128 | file | 2007-06-13 14:16:00 -0400 (Wed, - 13 Jun 2007) | 10 lines Merged revisions 69127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 - lines Return group counting to previous behavior where you could - only have one group per category. (issue #9711 reported by - irroot) ........ ................ - -2007-06-13 17:37 +0000 [r69081-69108] Jason Parker <jparker@digium.com> - - * res/res_config_pgsql.c: Continuation of issue 9968 (revision - 69081). This should be the last one. - - * main/pbx.c, channels/chan_sip.c: Fixes for ast_strlen_zero() - janitor project. Issue 9968, patch by eliel. - -2007-06-13 16:59 +0000 [r69017-69072] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 69071 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | - 3 lines Clarify a bit of logic. This doesn't change behavior in - any way, but it is helpful when following the logic to debug - problems like 9235. ........ - - * /, channels/chan_iax2.c: Merged revisions 69069 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69069 | russell | 2007-06-13 11:29:12 -0500 (Wed, 13 Jun 2007) | - 3 lines Fix a place where a chan_iax2 pvt struct was accessed - without the lock held. This issue was reported to me via email by - Dmitry Mishchenko. Thanks! ........ - - * res/snmp/agent.c: Simplify some logic and convert spaces to tabs - - * res/snmp/agent.c: The variable used for the return value must be - declared as static. I broke this when applying the patch, sorry! - (issue #9637, jeffg) - - * include/asterisk/logger.h: Put parenthesis around the level - argument to ast_debug() - - * /, cdr/cdr_pgsql.c: Merged revisions 69016 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69016 | russell | 2007-06-12 14:40:17 -0500 (Tue, 12 Jun 2007) | - 4 lines Fix a memory leak pointed out by prashant_jois in - #asterisk-bugs. PQclear() was not called on the result structure - after doing a PQexec(). Also, fix up some formatting in passing. - ........ - -2007-06-12 19:38 +0000 [r69013-69015] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 69014 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69014 | file | 2007-06-12 15:36:29 -0400 (Tue, 12 Jun 2007) | 2 - lines Change the full frame dropping log message to debug to - avoid future bug reports. ........ - - * /, channels/chan_iax2.c: Merged revisions 69012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69012 | file | 2007-06-12 15:26:38 -0400 (Tue, 12 Jun 2007) | 2 - lines Schedule the sending of a PING packet a second later than - previously so that it does not collide with the LAGRQ. ........ - -2007-06-12 19:19 +0000 [r68970-69011] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 69010 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | - 12 lines In ast_channel_make_compatible(), just return if the - channels' read and write formats already match up. There are code - paths that call this function on a pair of channels multiple - times. This made calls fail that were using g729 in some cases. - The reason is that codec_g729a will unregister itself from the - list of available translators will all licenses are in use. So, - the first time the function got called, the right translation - path was allocated. However, the second time it got called, the - code would not find a translation path to/from g729 and make the - call fail, even if the channel actually already had a g729 - translation path allocated. (SPD-32) ........ - - * main/pbx.c: Convert pbx.c to use ast_debug() for debug logging. - (issue #9925, dimas) - - * include/asterisk/logger.h: Add a new macro, ast_debug(), which - combines the check of the value of option_debug and the actual - call to ast_log(). (issue #9925, dimas) - - * doc/ast_appdocs.tex: update the dump of application docs - - * apps/app_dial.c, apps/app_privacy.c, apps/app_authenticate.c, - channels/chan_agent.c, apps/app_image.c, apps/app_chanisavail.c, - apps/app_transfer.c, apps/app_system.c, apps/app_queue.c, - apps/app_playback.c, apps/app_controlplayback.c, - apps/app_osplookup.c, apps/app_sendtext.c, apps/app_minivm.c, - apps/app_url.c, pbx/pbx_config.c, include/asterisk/options.h, - apps/app_voicemail.c: Completely remove all of the code related - to jumping to priority n + 101. yay! (issue #9926, caio1982) - -2007-06-12 14:26 +0000 [r68900-68923] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 68922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, - 12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 - lines Bring RTP back to Asterisk at the end of a native bridge no - matter what. ........ ................ - - * main/autoservice.c, main/app.c: Even more minor code cleanup! - - * main/channel.c: Minor code cleanup. - - * channels/chan_agent.c: Remove old stuff from the - AgentCallbackLogin days and only autocomplete agents in the agent - logoff CLI command that are logged in. (issue #9952 reported by - eliel) - -2007-06-11 22:31 +0000 [r68855] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/frame.c: corrected CLI 'core show codecs' syntax for issue - 9945, thanks eserra. - -2007-06-11 22:21 +0000 [r68854] Tilghman Lesher <tlesher@digium.com> - - * apps/app_disa.c, UPGRADE.txt: Issue 8971 - Allow DISA input to be - ended with a '#'. - -2007-06-11 22:07 +0000 [r68816-68831] Jason Parker <jparker@digium.com> - - * main/manager.c, configs/manager.conf.sample: Change - displayconnects option in manager.conf to be per-user. Issue - 9932, patch by eliel - - * /, include/asterisk/time.h: Merged revisions 68814 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r68814 | qwell | 2007-06-11 16:20:15 -0500 (Mon, 11 Jun - 2007) | 2 lines Solaris 10 sometimes (?) needs this include in - order to have NULL defined. ........ - -2007-06-11 20:51 +0000 [r68782] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_directory.c: Merged revisions 68781 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68781 | tilghman | 2007-06-11 15:45:53 -0500 (Mon, 11 Jun 2007) - | 2 lines Issue 9947 - fn2 was unused / incorrectly used ........ - -2007-06-11 17:05 +0000 [r68740] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 68733 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68733 | crichter | 2007-06-11 18:57:59 +0200 - (Mo, 11 Jun 2007) | 9 lines Merged revisions 68732 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 - Jun 2007) | 1 line added check for NULL Pointer when calling - misdn_new. Asterisk does not allow us to create channels anymore - when stop gracefully is used :). also modified the - restart_indicator to 0 ........ ................ - -2007-06-11 14:41 +0000 [r68662-68685] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Change channel list to read/write list... I'm - crazy. - - * main/channel.c, /: Merged revisions 68683 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68683 | file | 2007-06-11 10:33:12 -0400 (Mon, - 11 Jun 2007) | 10 lines Merged revisions 68682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 - lines Improve deadlock handling of the channel list. (issue #8376 - reported by one47) ........ ................ - - * main/manager.c: Add username completion for manager show user CLI - command. (issue #9929 reported by eliel) - - * configs/sip.conf.sample: Update documentation for proper CLI - commands. (issue #9936 reported by eserra) - -2007-06-11 11:40 +0000 [r68661] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - channels/misdn/isdn_lib.c: Merged revisions 68644 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68644 | crichter | 2007-06-11 12:29:18 +0200 - (Mo, 11 Jun 2007) | 9 lines Merged revisions 68631 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 - Jun 2007) | 1 line fixed problem that the dummybc chanels had no - lock, checking for the lock now. Also fixed the channel restart - stuff, we can now specify and restart particular channels too. - ........ ................ - -2007-06-11 04:28 +0000 [r68596] Tilghman Lesher <tlesher@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 68595 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68595 | tilghman | 2007-06-10 23:21:30 -0500 (Sun, 10 Jun 2007) - | 2 lines "dialplan save" produced garbage in the config file - ........ - -2007-06-09 01:06 +0000 [r68575] Jason Parker <jparker@digium.com> - - * channels/chan_misdn.c: Fix compile errors in chan_misdn.c - Reported by d1mas in #asterisk-bugs on IRC. - -2007-06-08 22:23 +0000 [r68473-68528] Russell Bryant <russell@digium.com> - - * /, apps/app_dictate.c: Merged revisions 68527 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68527 | russell | 2007-06-08 17:23:22 -0500 - (Fri, 08 Jun 2007) | 12 lines Merged revisions 68526 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 - Jun 2007) | 4 lines Don't automatically hang up after running - Dictate so that callers can exit cleanly using '#' (closes issue - #9577, patch from Thomas Andrews) ........ ................ - - * doc/asterisk-mib.txt, res/snmp/agent.c: Add support for - retrieving the number of channels that are currently bridged via - SNMP. (closes issue #9637, initial patch from jeffg, modified by - me) - - * include/asterisk/app.h, res/res_agi.c, main/app.c, - apps/app_controlplayback.c, apps/app_voicemail.c: Add an option - for ControlPlayback to be able to start at an offset from the - beginning of the file. Also, add a channel variable that - indicates the location in the file where the Playback was - stopped. (closes issue #7655, patch from sharkey) - - * main/pbx.c: Add channel variable manager event (issue #7291, - patch from tonyh and jontow) - -2007-06-08 16:03 +0000 [r68453] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 68450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68450 | kpfleming | 2007-06-08 10:52:47 -0500 (Fri, 08 Jun 2007) - | 2 lines actually remember the type/subclass of full frames that - are in process ........ - -2007-06-08 15:51 +0000 [r68449] Jason Parker <jparker@digium.com> - - * res/res_config_sqlite.c: Fix incorrect logic for param count. - Issue 9918. - -2007-06-08 15:32 +0000 [r68448] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Minor formatting change to test changes to - mantis auto-closing issues (closes issue #6000) - -2007-06-08 00:18 +0000 [r68374-68405] Joshua Colp <jcolp@digium.com> - - * /, main/say.c: Merged revisions 68401 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68401 | file | 2007-06-07 20:17:04 -0400 (Thu, - 07 Jun 2007) | 10 lines Merged revisions 68397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 - lines Don't call ast_waitstream_full when the control file - descriptor and audio file descriptor are not set, simply call - ast_waitstream! (issue #8530 reported by rickead2000) ........ - ................ - - * main/dnsmgr.c, /: Merged revisions 68370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68370 | file | 2007-06-07 20:02:34 -0400 (Thu, - 07 Jun 2007) | 10 lines Merged revisions 68368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 - lines Do a DNS lookup immediately upon calling the dnsmgr - function, don't wait until a refresh happens. (issue #9097 - reported by plack) ........ ................ - -2007-06-07 23:17 +0000 [r68339-68359] Russell Bryant <russell@digium.com> - - * /, main/say.c: Merged revisions 68354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68354 | russell | 2007-06-07 18:14:45 -0500 - (Thu, 07 Jun 2007) | 11 lines Merged revisions 68351 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 - Jun 2007) | 3 lines Fix a problem where saying a character - wouldn't properly break out when the caller pressed '#' (issue - #8113, reported by patbaker82, patch from jamesgolovich (hey, - long time no see!) and patbaker82) ........ ................ - - * include/asterisk/devicestate.h, channels/chan_sip.c, - contrib/asterisk-ng-doxygen, main/devicestate.c, - include/asterisk/manager.h, res/res_config_sqlite.c, main/rtp.c, - include/asterisk/http.h, include/asterisk/doxyref.h, - main/manager.c, include/asterisk/event.h, funcs/func_shell.c, - apps/app_skel.c, channels/chan_h323.c, - include/asterisk/strings.h, include/asterisk/stringfields.h: Fix - a bunch of doxygen errors and document more things (issue #9842, - snuffy) - -2007-06-07 23:00 +0000 [r68327] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 68326 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68326 | qwell | 2007-06-07 18:00:01 -0500 (Thu, 07 Jun 2007) | 5 - lines Fix incorrect French syntax of "old messages". Request for - feedback was sent to asterisk-dev mailing list, with little - response. Issue 9118, patch by junky. ........ - -2007-06-07 22:38 +0000 [r68325] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a couple of places that got missed in - the conversion to using the new API call for creating detached - threads. (issue #9915, reported by elguro, fixed by me) - -2007-06-07 22:18 +0000 [r68321] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 68313 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68313 | kpfleming | 2007-06-07 17:14:35 -0500 (Thu, 07 Jun 2007) - | 6 lines some improvements to the IAX2 full frame dropping logic - recently added: - use inaddrcmp(), since we have it - output the - type of frame and subclass being dropped, and the type/subclass - that is already being processed (which caused the drop) ........ - -2007-06-07 21:22 +0000 [r68284-68289] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: Doxygenify a lot of the functions in res_jabber - (issue #9886, snuffy) - - * /, channels/chan_agent.c, apps/app_queue.c: Merged revisions - 68280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68280 | russell | 2007-06-07 16:16:07 -0500 (Thu, 07 Jun 2007) | - 4 lines Fix loading persistent queue members when using realtime - configuration for queues. Also, remove an unneeded leading slash - for the astdb family. (issue #9911, patch by atis) ........ - -2007-06-07 20:25 +0000 [r68220-68251] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 68249 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r68249 | qwell | 2007-06-07 15:25:18 -0500 (Thu, 07 Jun - 2007) | 4 lines Fix an issue with newer phones which require - packets be padded out to the correct length. Issue 9887, patch by - DEA. ........ - - * /, apps/app_voicemail.c: Merged revisions 68211 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68211 | qwell | 2007-06-07 15:06:00 -0500 (Thu, - 07 Jun 2007) | 12 lines Merged revisions 68204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 - lines Don't try to save voicemail greetings unless the user - presses '1' to accept/save. Issue 9904, patch by me. ........ - ................ - -2007-06-07 19:51 +0000 [r68201] Olle Johansson <oej@edvina.net> - - * CREDITS: Adding Philippe to CREDITS for hard work on detecting - bugs in our jabber/jingle integration - -2007-06-07 19:50 +0000 [r68200] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 68198 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun - 2007) | 5 lines Submitting a fix for Issue 8016. Added a check to - make sure that greetings get stored properly. (Issue 8016, - reported by edhorton, patched by alamantia with modification by - me. Thanks to Jason Parker for the advice on this). ........ - -2007-06-07 19:49 +0000 [r68195-68199] Olle Johansson <oej@edvina.net> - - * /, channels/chan_features.c: Merged revisions 68196 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r68196 | oej | 2007-06-07 21:46:10 +0200 (Thu, 07 Jun - 2007) | 2 lines Disable chan_features by default in menuselect - ........ - - * channels/chan_sip.c: - Doxygen updates - Adding docs on flags to - be able to clean up a bit - -2007-06-07 19:31 +0000 [r68193] Russell Bryant <russell@digium.com> - - * /, main/strcompat.c: Merged revisions 68192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68192 | russell | 2007-06-07 14:30:30 -0500 (Thu, 07 Jun 2007) | - 3 lines Include stdarg.h for build issues on Solaris (issue - #9381) ........ - -2007-06-07 18:41 +0000 [r68138-68158] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 68157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2 - lines Fix logic when doing a name based channel search for a - structure when you want to start from a specific point in the - channel list. (issue #9324 reported by slavon) ........ - - * doc/queues-with-callback-members.tex: AEL in trunk now uses GOSUB - so we have to update the queues with callback members example. - (issue #9813 reported by Mike Anikienko) - -2007-06-07 15:48 +0000 [r68118] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: Minor formatting change ... testing mantis - stuff to see if we're done (issue #9790) (closes issue #9816) - -2007-06-07 14:23 +0000 [r68072] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 68071 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r68071 | file | 2007-06-07 10:21:59 -0400 (Thu, - 07 Jun 2007) | 10 lines Merged revisions 68070 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 - lines Allow the 'g' option to work if used with the 'S' option. - (issue #9888 reported by gasparz) ........ ................ - -2007-06-07 10:06 +0000 [r67991-68040] Olle Johansson <oej@edvina.net> - - * /, res/res_jabber.c: Merged revisions 68030 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68030 | oej | 2007-06-07 12:00:17 +0200 (Thu, 07 Jun 2007) | 2 - lines Adding a few Todo's to res_jabber so we don't forget. - ........ - - * /, res/res_jabber.c: Merged revisions 68028 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68028 | oej | 2007-06-07 11:55:13 +0200 (Thu, 07 Jun 2007) | 4 - lines Ok, we found out that this is not about if you have any - *active* clients using TLS, but if you have initialized TLS at - all during the lifetime of the module. So if you reload to - disable TLS, it won't help. ........ - - * /, res/res_jabber.c: Merged revisions 68027 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r68027 | oej | 2007-06-07 11:42:26 +0200 (Thu, 07 Jun 2007) | 8 - lines If you have a jabber client that uses TLS, refuse unload. - Bad fix, but will prevent crashes while we are trying to find a - workaround. Iksemel development seems to have stalled and we - might have to stop using the TCP/TLS connections in that library - and use our own, which would scale better from a poll/select - perspective I guess. It would also make it easier to migrate to - OpenSSL and stop Asterisk from depending on both OpenSSL and - GnuTLS. ........ - - * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions - 67993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 - lines Issue #9738 - Make sure we can unload res_jabber. Patch by - phsultan - thanks! Due to a bug in the iksemel library, this will - not work if you are using GTLS in the connection. That's being - investigated. If you figure out a way to handle that without us - having to patch iksemel, let us know in the bug report. Thanks. - ........ - - * res/res_jabber.c: Simplification of res_jabber code (done at - Inria, Paris with Philippe) - - * main/strcompat.c: Reverting part of #67864 to be able to compile - agi/eagi-test that relies on this without having ast_log and - other asterisk api functions available - I could not compile on - OS/X without reverting this. - -2007-06-07 00:12 +0000 [r67925-67944] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 67941 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67941 | file | 2007-06-06 20:10:48 -0400 (Wed, - 06 Jun 2007) | 10 lines Merged revisions 67938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 - lines Only notify the devicestate system of a peer state change - when the peer is built from the config file. (issue #9900 - reported by arkadia) ........ ................ - - * /, main/file.c: Merged revisions 67924 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67924 | file | 2007-06-06 19:38:15 -0400 (Wed, 06 Jun 2007) | 2 - lines Properly handle cases where a stream can't be written to. - (issue #9757 reported by junky) ........ - -2007-06-06 23:12 +0000 [r67920] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Allow overlapdialing directions to be - configurable. Bug #8554 - -2007-06-06 22:35 +0000 [r67901] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_iax2.c: added CLI 'iax2 unregister <peername>' for - issue 9812, thanks eliel - -2007-06-06 22:27 +0000 [r67875-67895] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: Remove our little - joke that was making fun of email disclaimers which nobody else - seemed to think was very funny. Oh well ... :) - - * /, res/res_snmp.c: Merged revisions 67872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67872 | russell | 2007-06-06 17:08:02 -0500 (Wed, 06 Jun 2007) | - 6 lines Disable reload functionality in res_snmp. It is not - possible to initialize the snmp library more than once without - completely unloading the module and loading it again. (issue - #9571, reported by hristo, additional helpful debug information - from festr, patch from me) ........ - -2007-06-06 21:20 +0000 [r67864] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, main/autoservice.c, main/frame.c, - channels/chan_local.c, apps/app_readfile.c, res/res_features.c, - main/threadstorage.c, main/say.c, funcs/func_strings.c, - apps/app_alarmreceiver.c, main/devicestate.c, - cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, - main/indications.c, main/config.c, main/loader.c, main/cli.c, - res/res_smdi.c, channels/chan_skinny.c, main/strcompat.c, - main/http.c, apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, - res/res_speech.c, apps/app_milliwatt.c, main/sched.c, - apps/app_dial.c, main/pbx.c, channels/chan_agent.c, - channels/iax2-provision.c, channels/iax2-parser.c, - main/chanvars.c, res/res_monitor.c, main/netsock.c, - apps/app_speech_utils.c, channels/chan_misdn.c, - funcs/func_curl.c, main/fixedjitterbuf.c, apps/app_macro.c, - res/res_indications.c, apps/app_mixmonitor.c, main/asterisk.c, - res/res_odbc.c, main/dlfcn.c, apps/app_voicemail.c, - channels/chan_vpb.cc, apps/app_meetme.c, main/utils.c, - res/res_musiconhold.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, - apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c, - res/res_config_sqlite.c, main/enum.c, channels/misdn_config.c, - main/io.c, main/channel.c, main/cdr.c, funcs/func_enum.c, - main/dial.c, main/manager.c, apps/app_osplookup.c, main/tdd.c, - funcs/func_odbc.c, cdr/cdr_sqlite.c, res/res_agi.c, - apps/app_minivm.c, main/app.c, apps/app_directory.c, - apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, - codecs/codec_lpc10.c, res/res_config_pgsql.c, - channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c, - apps/app_festival.c, main/translate.c, main/jitterbuf.c, - main/acl.c, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, - cdr/cdr_tds.c, main/file.c, main/callerid.c, main/event.c, - funcs/func_devstate.c, funcs/func_callerid.c, main/dsp.c: Issue - 9869 - replace malloc and memset with ast_calloc, and other - coding guidelines changes - -2007-06-06 21:16 +0000 [r67813-67863] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 67862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | - 4 lines Fix a crash when doing call pickups with SIP phones. The - code unlocked the channel when it should not have. (issue #9652, - reported by corruptor, fixed by me) ........ - - * res/res_features.c, include/asterisk/features.h: Constify the - return values of ast_parking_ext() and ast_pickup_ext() - - * main/manager.c: Minor formatting change to test closing mantis - issues through commit tags (closes issue #9828) - - * main/manager.c: Minor formatting change to test closing mantis - issues through commit tags (closes issue #9828) - - * apps/app_voicemail.c: Please forgive this flood of tiny changes - ... this will be cool when it works how we want it to :) (testing - mantis+svn) (issue #9828) - -2007-06-06 19:46 +0000 [r67808] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 67804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun - 2007) | 10 lines Fix for Issue 9810. There was a segfault under a - specific set of circumstances: 1. VoiceMailMain was configured in - the dialplan with an extension as its argument 2. A message was - left for this mailbox 3. Tried to call VoiceMailMain but hung up - before entering password. This was fixed by checking that a - pointer was non-null prior to trying to dereference it. (Issue - 9810, reported by xmarksthespot, patched by Corydon76 with - modifications by me). ........ - -2007-06-06 19:44 +0000 [r67787-67807] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: minor formatting change ... testing - mantis/svn (issue #9828) - - * apps/app_voicemail.c: Don't try to check the result of alloca ... - ... testing mantis/svn stuff ... (issue #9828) - - * main/dsp.c: Yet another minor change to test mantis/svn (issue - #9828) - - * main/dsp.c: minor formatting change ... testing mantis/svn (issue - #9828) - - * main/dsp.c: minor formatting change ... testing mantis/svn (issue - #9828) - - * main/app.c: Formatting change ... testing (issue #9828) - -2007-06-06 19:02 +0000 [r67784] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing a crash wherein Asterisk would - segfault when attempting to leave a voicemail when IMAP storage - was enabled. Though no bug was reported to the bugtracker, there - was mention of this made as a note on bug 9810 by edhorton. - -2007-06-06 19:00 +0000 [r67697-67782] Russell Bryant <russell@digium.com> - - * main/app.c: Make another formatting change ... testing mantis/svn - stuff (issue #9828) - - * main/app.c: Another minor formatting change ... testing - mantis/svn (issue #9828) - - * main/app.c: Minor formatting change ... testing mantis/svn (issue - #9828) - - * channels/chan_iax2.c: Make another small tweak ... mantis/svn - testing (issue #9828) - - * res/res_features.c: Another tiny formatting change for testing - ... (issue #9828) - - * main/app.c: More random formatting changes to test Mantis/SVN - integration (issue #9828) - - * main/app.c: Make a completely arbitrary formatting change to test - out some Mantis/SVN integration stuff. (issue #9828) - - * main/channel.c, /, include/asterisk/linkedlists.h: Merged - revisions 67716 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67716 | russell | 2007-06-06 11:55:59 -0500 - (Wed, 06 Jun 2007) | 13 lines Merged revisions 67715 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 - Jun 2007) | 5 lines We have some bug reports showing crashes due - to a double free of a channel. Add a sanity check to - ast_channel_free() to make sure we don't go on trying to free a - channel that wasn't found in the channel list. (issue #8850, and - others...) ........ ................ - - * res/res_features.c: Change "show parkedcalls" to "parkedcalls - show" and mark the previous command as deprecated. Also, convert - the CLI command to the new style. (issue #9861, patch from eliel) - -2007-06-06 13:32 +0000 [r67595-67651] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 67650 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, - 06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 - lines Reinvite the RTP back to the Asterisk machine when the - timeout happens. (issue #9888 reported by gasparz) ........ - ................ - - * /, main/translate.c: Merged revisions 67631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67631 | file | 2007-06-06 09:18:39 -0400 (Wed, 06 Jun 2007) | 2 - lines Fix plc_samples warning when registering a translator. - (issue #9897 reported by xylome) ........ - - * /, apps/app_directed_pickup.c: Merged revisions 67626 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67626 | file | 2007-06-06 09:16:34 -0400 (Wed, 06 Jun 2007) | 2 - lines Include macroexten while searching for a channel to pick up - in case they are in a macro. (issue #9491 reported by jamesb63) - ........ - - * /, res/res_agi.c: Merged revisions 67597 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67597 | file | 2007-06-06 08:34:06 -0400 (Wed, 06 Jun 2007) | 2 - lines Make the new "agi debug off" CLI command work. (issue #9890 - reported by eliel) ........ - - * channels/chan_zap.c: When SS7 is enabled add w/SS7 to the end. - (issue #9893 reported by Mike Anikienko) - - * /, main/devicestate.c: Merged revisions 67594 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67594 | file | 2007-06-06 08:20:27 -0400 (Wed, - 06 Jun 2007) | 10 lines Merged revisions 67593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 - lines Revert channel name splitting fix for Zap. The moral of the - story is don't use - in your user/peer names. (issue #9668 - reported by stevedavies) ........ ................ - -2007-06-05 23:02 +0000 [r67560] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 67558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | - 5 lines Fix some crashes related to the use of the "meetme" CLI - command. The code for this command was not locking the conference - list at all. (issue #9351, reported by and patch submitted by - Junk-Y, committed patch is different and by me) ........ - -2007-06-05 22:59 +0000 [r67557] Mark Michelson <mmichelson@digium.com> - - * main/cli.c: Found a bug where when "core set debug #" is used, - the verbosity is read as the old value instead of the old debug - value, leading to an erroneous status message after setting. This - was purely a cosmetic issue and had no other underlying problems. - -2007-06-05 22:04 +0000 [r67529] Steve Murphy <murf@digium.com> - - * utils/Makefile, /, pbx/ael/ael.tab.c, pbx/ael/ael.y, - pbx/pbx_ael.c, pbx/Makefile: Merged revisions 67526 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r67526 | murf | 2007-06-05 15:30:18 -0600 (Tue, 05 Jun - 2007) | 1 line this fixes bug 9883, wherein macros were not - allowing the includes construct. fixed and tested, looks OK. Now - includes can serve as an adjunct to catch. ........ - -2007-06-05 20:55 +0000 [r67493] Russell Bryant <russell@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 67492 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) | - 16 lines This bug has been hanging over my head ever since I - wrote this SLA code. Every time I tried to go debug it by adding - some debug output, the behavior would change. It turns out I - wasn't crazy. I had the following piece of code: if (remove) - AST_LIST_REMOVE_CURRENT(...); Well, AST_LIST_REMOVE_CURRENT was - not wrapped in braces, so my conditional statement didn't do much - good at all. It always ran at least all of the macro minus the - first statement, so I was seeing list entries magically disappear - when they weren't supposed to. After many hours of debugging, I - have come to this extremely irritating fix. :) (issues #9581, - #9497) ........ - -2007-06-05 20:16 +0000 [r67486] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Merged revisions 67424 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun - 2007) | 5 lines Fix for bug number 9786, wherein voicemails saved - to IMAP storage using extensions other than gsm were unable to be - played over the phone. (Issue 9786, reporter: xmarksthespot, - Patched by xmarksthe spot with revisions by me, reviewed by - Russell Bryant). ........ - -2007-06-05 19:50 +0000 [r67458] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 67457 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67457 | russell | 2007-06-05 14:48:02 -0500 (Tue, 05 Jun 2007) | - 2 lines Suppress a bunch of debug output unless option_debug is - on ........ - -2007-06-05 18:23 +0000 [r67423] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 67420 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67420 | murf | 2007-06-05 12:17:28 -0600 (Tue, 05 Jun 2007) | 1 - line Added code to automatically add a default case to switches - that don't have one. In some cases, rather than fall thru, it - results in a goto with -1 result, which terminates the extension; - a sort of dialplan seqfault, sort of. This was required to fix - bug reported in 9881 ........ - -2007-06-05 18:19 +0000 [r67398-67422] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 67421 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r67421 | qwell | 2007-06-05 13:18:24 -0500 (Tue, 05 Jun - 2007) | 4 lines Correctly update date/time on devices throughout - the life of the device, instead of just at registration. Issue - 9152, yet another patch by DEA. ........ - - * main/manager.c: Make sure we default allowmultiplelogin to - on/yes, per the default stated in the config. Issue 9885, patch - by eliel. - -2007-06-05 17:24 +0000 [r67397] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/misdn/isdn_msg_parser.c: changed #if DEBUG to #ifdef - DEBUG to fix make failure when configured with --enable-dev-mode - -2007-06-05 17:11 +0000 [r67361-67380] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Improve the way that the zaptel channel name - is created by using the Asterisk strings API and by only - allocating space on the stack - - * /: Merged revisions 67360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67360 | russell | 2007-06-05 11:56:36 -0500 (Tue, 05 Jun 2007) | - 5 lines Fix a problem that showed itself by causing Zap channel - names to be completely bogus on my machine. - ast_safe_string_alloc() was broken. It called vsnprintf() on a - va_args list twice without re-initializing it. After the first - usage, va_end() and va_start() must be called again. ........ - -2007-06-05 16:21 +0000 [r67345-67350] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/chan_misdn_config.h: Merged revisions 67334 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67334 | crichter | 2007-06-05 18:14:07 +0200 - (Di, 05 Jun 2007) | 9 lines Merged revisions 67307 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 - Jun 2007) | 1 line briding is a bool, fixed copy and paste issue. - ........ ................ - - * channels/chan_misdn.c, /: Merged revisions 67329 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67329 | crichter | 2007-06-05 18:11:57 +0200 - (Di, 05 Jun 2007) | 9 lines Merged revisions 67306 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 - Jun 2007) | 1 line simplified the EVENT_SETUP handling in the - cb_events function a lot. Commented the different possibilities a - bit and made functions of shared code. When the dialed extension - does not exist in the extensions.conf we'll jump into the 'i' - extension if this does exist, else we disconnect the call with - the cause:1 = No Route to Destination. ........ ................ - -2007-06-05 15:54 +0000 [r67310] Russell Bryant <russell@digium.com> - - * /, include/asterisk/module.h, main/asterisk.c, main/loader.c: - Merged revisions 67308 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) | - 5 lines When shutting down "gracefully", go through and run the - unload() callbacks for all of the modules. "stop now" is - considered a non-graceful shutdown and will not go through this - process. (issue #9804, reported by chrisost, patch by me) - ........ - -2007-06-05 15:24 +0000 [r67305] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 67304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67304 | file | 2007-06-05 12:22:30 -0300 (Tue, 05 Jun 2007) | 2 - lines Only muck with the thread structure if an idle one was - found/created. ........ - -2007-06-05 14:59 +0000 [r67272-67273] Russell Bryant <russell@digium.com> - - * doc/CODING-GUIDELINES: add a note about inline comments - - * channels/chan_iax2.c: Doxygenify the comments for new members of - the iax2_thread struct - -2007-06-05 14:45 +0000 [r67271] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 67270 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007) - | 3 lines ensure that a burst of full frames (AST_FRAME_DTMF - being the prime example) will not be processed out of order... - this is a brute force fix, but seems to be the safest fix for now - (thanks to the Digium PQ department for finding this bug) - ........ - -2007-06-05 11:48 +0000 [r67240] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, - channels/misdn_config.c: Merged revisions 67210 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67210 | crichter | 2007-06-05 12:25:32 +0200 - (Di, 05 Jun 2007) | 9 lines Merged revisions 67209 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 - Jun 2007) | 1 line added possibility to deactivate bridging per - port ........ ................ - -2007-06-04 23:45 +0000 [r67164] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_math.c: Merged revisions 67162 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67162 | tilghman | 2007-06-04 18:43:01 -0500 - (Mon, 04 Jun 2007) | 10 lines Merged revisions 67161 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 - Jun 2007) | 2 lines According to MATH, 0+1181000386 = 1181000448. - Oops. ........ ................ - -2007-06-04 23:32 +0000 [r67160] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 67158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) | - 5 lines Fix up a bunch of places where the iax2 pvt structure can - disappear and the code did not account for it and crashes. - (issues #9642, #9569, #9666, probably others ... based on the - work by stevedavies and mihai, with additional changes from me) - ........ - -2007-06-04 23:29 +0000 [r67122-67157] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 67156 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun - 2007) | 6 lines Fix for skinny keepalives. If there is no traffic - from the phone for (keep_alive * 1100) ms (arbitrarily adding 10% - for network issues, etc), unregister the device. Issue 8394, - patch by DEA. ........ - - * /, channels/chan_mgcp.c: Merged revisions 67121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4 - lines Fixes for dtmf/dialing with mgcp (similar to the recent fix - for chan_skinny) Issue 9855, patch by DEA. ........ - -2007-06-04 22:29 +0000 [r67120] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 67119 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) | - 6 lines Add comments for two functions that get called with the - appropriate call locked, but perform operations that could result - in the pvt structure getting destroyed before returning again, - causing numerous seg faults all over the module. (inspired by - issues #9642, #9569, and #9666, and the work done by stevedavies - and mihai) ........ - -2007-06-04 22:15 +0000 [r67095] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 67073 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67073 | murf | 2007-06-04 15:59:34 -0600 (Mon, 04 Jun 2007) | 1 - line This typo has been here since 1.4 forked. It has been the - source of heartburn to many a dialplan/CDR programmer. ........ - -2007-06-04 21:48 +0000 [r67070-67072] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: Merged revisions 67071 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | - 2 lines Add a missing \n. (pointed out by jcmoore on IRC) - ........ - - * channels/chan_iax2.c: Remove a leftover unlock and lock of the - iax2 pvt struct lock that was left over from my attempt at - putting pvt structs in a hash table. It can cause seg faults, and - has no reason to stay. (issue #9642, pointed out by stevedavies) - -2007-06-04 19:32 +0000 [r67063-67069] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 67068 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 - lines Better handle SIP devices that say they have SDP content... - but really don't. (issue #9398 reported by mthomasslo) ........ - - * apps/app_dial.c, /: Merged revisions 67066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 - lines Initialize cidname variable to nothing since it may be used - without having been touched. (issue #9661 reported by dimas) - ........ - - * /, res/res_features.c: Merged revisions 67064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67064 | file | 2007-06-04 13:41:59 -0400 (Mon, 04 Jun 2007) | 2 - lines Returning a value that indicates the parking of a call was - a success when it really wasn't (because the parking slot - selected was in use) is the wrong thing to do. (issue #9723 - reported by mdu113) ........ - - * apps/app_directed_pickup.c: Minor clean up. Constify a few - variables and use ast_strlen_zero in a few places. - -2007-06-04 17:12 +0000 [r67062] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.mandrake.asterisk, /, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.gentoo.asterisk, - contrib/init.d/rc.mandrake.zaptel, - contrib/init.d/rc.slackware.asterisk: Merged revisions 67061 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r67061 | tilghman | 2007-06-04 12:11:43 -0500 - (Mon, 04 Jun 2007) | 10 lines Merged revisions 67060 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 - Jun 2007) | 2 lines Add revision Id tags (by request of tzafrir) - ........ ................ - -2007-06-04 16:03 +0000 [r67024-67029] Russell Bryant <russell@digium.com> - - * /, configure, configure.ac: Merged revisions 67026 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r67026 | russell | 2007-06-04 11:02:31 -0500 (Mon, 04 - Jun 2007) | 6 lines Change the configure script to build a test - program against libcurl to make sure the results from curl-config - can be used to compile successfully. This is intended to help - prevent a situation where you are cross compiling, and the - configure script finds the curl library installed on the host. - (issue #9865, reported and patched by zandbelt) ........ - - * main/ast_expr2f.c, pbx/ael/ael_lex.c, main/app.c: Change javadoc - style code documentation to the same format we use elsewhere. - (issue #9864, patch from snuffy) - -2007-06-04 15:53 +0000 [r67023] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_jabber.c: Merged revisions 67021 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67021 | tilghman | 2007-06-04 10:50:16 -0500 (Mon, 04 Jun 2007) - | 2 lines Issue 9739 - Malformed jid causes a crash ........ - -2007-06-04 15:50 +0000 [r67016-67022] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 67020 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) | - 7 lines Resolve a deadlock in chan_iax2. When handling an - implicit ACK to a frame that was marked as the final transmission - for a call, don't call iax2_destroy() for that call while the - global frame queue is still locked. There is a very nice - explanation of the deadlock in the report. (issue #9663, thorough - report and patch from stevedavies, additional positive test - reports from mihai and joff_oconnell) ........ - - * include/asterisk/stringfields.h: Fix some compiler warnings in - C++ modules. (issue #9866, reported by osk, patch by Corydon76) - - * channels/chan_sip.c, main/netsock.c: Fix a couple of places where - "tos" was used instead of "cos". (issue #9540, patch by IgorG) - -2007-06-04 11:48 +0000 [r66998] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c: Add support for autocompleting start/stop - options of the mixmonitor CLI command. (issue #9862 reported by - eliel) - -2007-06-03 06:10 +0000 [r66981] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_jingle.c, channels/chan_phone.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_gtalk.c, channels/chan_nbs.c, channels/chan_mgcp.c: - ast_calloc janitor (Inspired by issue 9860) - -2007-06-01 23:39 +0000 [r66957-66959] Russell Bryant <russell@digium.com> - - * main/pbx.c: remove a bogus comment that came from copy/paste - - * include/asterisk/devicestate.h, include/asterisk.h, main/pbx.c, - include/asterisk/event_defs.h, main/devicestate.c, - include/asterisk/pbx.h, apps/app_queue.c, main/asterisk.c: Merge - major changes to the way device state is passed around Asterisk. - The two places that cared about device states were app_queue and - the hint code in pbx.c. The changes include converting it to use - the Asterisk event system, as well as other efficiency - improvements. * app_queue: This module used to register a - callback into devicestate.c to monitor device state changes. Now, - it is just a subscriber to Asterisk events with the type, device - state. * pbx.c hints: Previously, the device state processing - thread in devicestate.c would call ast_hint_state_changed() each - time the state of a device changed. Then, that code would go - looking for all the hints that monitor that device, and call - their callbacks. All of this blocked the device state processing - thread. Now, the hint code is a subscriber of Asterisk events - with the type, device state. Furthermore, when this code receives - a device state change event, it queues it up to be processed by - another thread so that it doesn't block one of the event - processing threads. - - * channels/chan_iax2.c: Remove 80 bytes in the iax2_registry struct - that weren't being used - -2007-06-01 21:49 +0000 [r66920] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_odbc.c: Merged revisions 66919 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66919 | tilghman | 2007-06-01 16:45:44 -0500 (Fri, 01 Jun 2007) - | 2 lines On some drivers, deallocating the statement handle - isn't enough. We also have to clear the cursor (nice, Oracle) - ........ - -2007-06-01 21:33 +0000 [r66910-66918] Mark Michelson <mmichelson@digium.com> - - * /: Merged revisions 66916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - ........ - - * /, apps/app_voicemail.c: Merged revisions 66897 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun - 2007) | 3 lines Submitting a fix for voicemail with IMAP storage. - Attachments with format specified as gsm were duplicated (i.e. - two attachments) were left. Thank you very much to xmarksthespot - for submitting the patch that fixed this. (Issues 9787 and 8873, - Reported by xmarksthespot and jerjer, patched by xmarksthespot) - ........ - -2007-06-01 19:42 +0000 [r66880-66882] Russell Bryant <russell@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 66881 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01 - Jun 2007) | 6 lines Changes to the way DTMF is handled in the - core broke dialing in chan_skinny. This patch makes chan_skinny - usable again. I did not end up testing this, but there are - multiple positive test reports listed in the bug report. (issue - #9596, reported by pj, testing by pj and mvanbaak, and the fix - was written by DEA) ........ - - * /, apps/app_page.c: Merged revisions 66879 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) | - 2 lines List app_meetme as a module that app_page depends on. - ........ - -2007-06-01 18:36 +0000 [r66878] Jason Parker <jparker@digium.com> - - * res/res_config_sqlite.c: Documentation fixes for - res_config_sqlite. Issue 9854, patch by tzafrir. - -2007-06-01 13:48 +0000 [r66856] Russell Bryant <russell@digium.com> - - * configs/sip.conf.sample: Add some more information about the SIP - Disclaimer header. - -2007-05-31 23:04 +0000 [r66822] Tilghman Lesher <tlesher@digium.com> - - * /, doc/asterisk.8: Merged revisions 66821 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66821 | tilghman | 2007-05-31 18:03:28 -0500 (Thu, 31 May 2007) - | 2 lines Issue 9850 - update preferred command line syntax - ........ - -2007-05-31 21:23 +0000 [r66772-66818] Russell Bryant <russell@digium.com> - - * configs/sip.conf.sample: fix a typo. - - * channels/chan_sip.c, configs/sip.conf.sample: To satisfy some - legal concerns, add an option for chan_sip to include a - disclaimer along with SIP messages in the header, X-Disclaimer. - This is off by default. Also, the text of the disclaimer can be - customized in sip.conf. - - * include/asterisk/app.h, /, include/asterisk/speech.h, - res/res_speech.c: Merged revisions 66775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66775 | russell | 2007-05-31 13:41:58 -0500 (Thu, 31 May 2007) | - 3 lines Change a couple of header files to not use "new", which - is a reserved keyword in C++. (issue #9830, reported by osk) - ........ - - * res/res_features.c, CHANGES, configs/features.conf.sample: Add - support for configuring named groups of custom call features in - features.conf. This allows you to create a feature one time, and - then map it into groups for various different key mappings for - the same feature, as well as easy access control to groups of - features. (patch from bbryant) - - * res/res_features.c, configs/features.conf.sample: Revert changes - that snuck in with revision 66724. - - * apps/app_minivm.c: - Don't check if the list is empty needlessly - - Don't free structures before calling load_config(), because - load_config() already does it - Use the existing functions for - freeing the minivm structures instead of replicating the code - (issue #9846, patch from eliel) - -2007-05-31 17:16 +0000 [r66771] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_macro.c: Merged revisions 66770 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r66770 | tilghman | 2007-05-31 12:15:09 -0500 - (Thu, 31 May 2007) | 10 lines Merged revisions 66744 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 - May 2007) | 2 lines Issue 9818 - Fix for issue 8329 breaks - pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, - but only because we lack core API to do it. ........ - ................ - -2007-05-31 16:18 +0000 [r66769] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 66768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, - 31 May 2007) | 10 lines Merged revisions 66764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 - lines It is now possible for this path of execution to have the - frame pointer be NULL, therefore we need to check for it before - trying to access it. (issue #9836 reported by barthpbx) ........ - ................ - -2007-05-31 15:05 +0000 [r66734] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample, funcs/func_odbc.c: Issue 9799 - - Multirow results for func_odbc - -2007-05-31 14:52 +0000 [r66724] Russell Bryant <russell@digium.com> - - * res/res_features.c, apps/app_minivm.c, - configs/features.conf.sample: Fix a crash on reload by using - calloc() instead of malloc() to ensure that data is properly - initialized. (issue #9765, reported by MatsK, patch from eliel) - -2007-05-31 10:26 +0000 [r66705] Olle Johansson <oej@edvina.net> - - * include/asterisk/app.h, apps/app_osplookup.c, - include/asterisk/event.h, apps/app_meetme.c, channels/chan_sip.c, - include/asterisk/event_defs.h, apps/app_skel.c, - apps/app_minivm.c, res/res_jabber.c: Issue #9842 - Doxygen - updates by snuffy. Thanks! (Committed from Media Plaza in - Utrecht, Netherlands - Open Source VoIP conference) - -2007-05-30 23:44 +0000 [r66672] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 66671 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66671 | mmichelson | 2007-05-30 18:26:39 -0500 (Wed, 30 May - 2007) | 2 lines Fixed seg-faults when recording greetings in - voicemail with IMAP enabled. (Issue No. 9734, reported by - xmarksthespot, patched by me) ........ - -2007-05-30 17:23 +0000 [r66603-66638] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, channels/chan_features.c: This concludes my - tweaking of things. - -2007-05-30 05:17 +0000 [r66539-66585] Tilghman Lesher <tlesher@digium.com> - - * apps/app_channelredirect.c, channels/chan_vpb.cc, - res/res_config_odbc.c, funcs/func_shell.c, funcs/func_cdr.c, - apps/app_zapras.c, res/res_indications.c, apps/app_transfer.c, - apps/app_stack.c, funcs/func_devstate.c, res/res_config_sqlite.c, - res/res_odbc.c: Issue 9477 - Improve menuselect labels - - * /, funcs/func_strings.c: Merged revisions 66538 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r66538 | tilghman | 2007-05-29 16:56:07 -0500 - (Tue, 29 May 2007) | 10 lines Merged revisions 66537 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 - May 2007) | 2 lines If the value of a variable passed to FIELDQTY - is blank, then FIELDQTY should return 0, not 1. ........ - ................ - - * funcs/func_enum.c: Shorten description to a much more reasonable - length - -2007-05-29 19:53 +0000 [r66502-66505] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: oops. Thanks Terry. - - * /, channels/chan_sip.c: Merged revisions 66503 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 - lines Properly handle 408 request timeout - according to the RFC, - the dialog dies if a request in a dialog gets this response. - ........ - - * /, channels/chan_sip.c: Merged revisions 66474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 - lines Don't issue hangup on hangup on hangup on hangup (for - jcmoore) ........ - -2007-05-29 19:00 +0000 [r66471] Doug Bailey <dbailey@digium.com> - - * main/dsp.c: Changed the dtmf detection to integer based goertzel - algorithm. - -2007-05-29 16:46 +0000 [r66438] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 66437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 - lines Handle cases where a frame may have no data. (issue #9519 - reported by dmb) ........ - -2007-05-29 16:19 +0000 [r66432-66433] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 66414 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 - lines Don't reset hangupcause if we already have one ........ - - * /, channels/chan_sip.c: Merged revisions 66404 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 - lines Tracking down hanging channels, killing them one by one. - Issue #9235 and related ........ - -2007-05-29 15:44 +0000 [r66399] Joshua Colp <jcolp@digium.com> - - * /, doc/datastores.txt: Merged revisions 66398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66398 | file | 2007-05-29 11:43:16 -0400 (Tue, 29 May 2007) | 2 - lines Update datastores documentation. (issue #9801 reported by - mnicholson) ........ - -2007-05-29 10:02 +0000 [r66367] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 66363 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, - 29 May 2007) | 10 lines Merged revisions 66349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 - lines Issue #9802 - Change inuse counter on CANCEL ........ - ................ - -2007-05-28 23:28 +0000 [r66313-66315] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't try to unregister a peer using the sip - unregister CLI command if they are not registered. (issue #9811 - reported by eliel) - - * channels/chan_sip.c: Due to the way stringfields work the value - of the url pointer will always be non-NULL so we have to use - ast_strlen_zero to make sure it is not empty. (issue #9821 - reported by pj) - -2007-05-28 18:50 +0000 [r66295] Olle Johansson <oej@edvina.net> - - * apps/app_voicemail.c: - Don't re-invent existing headers (some - already existed in chan_sip) - Rename command so taht module name - comes first - -2007-05-28 15:59 +0000 [r66278] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_iconv.c (added): Issue 7021 - Add ICONV function for - converting between character sets - -2007-05-26 19:35 +0000 [r66225] Joshua Colp <jcolp@digium.com> - - * apps/app_minivm.c: Unlock the minivmlock when no configuration is - found. (issue #9814 reported by eliel) - -2007-05-26 06:07 +0000 [r66208] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Since this code now uses the API call for - creating a detached thread, there is no reason to keep a thread - attribute structure on the conference structure. (Pointed out by - Tony Mountifield on the asterisk-dev list) - -2007-05-25 15:08 +0000 [r66175-66178] Kevin P. Fleming <kpfleming@digium.com> - - * /: block change that is already here - - * channels/chan_jingle.c, configure, configure.ac: more minor fixes - -2007-05-25 14:49 +0000 [r66161] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c: Merged revisions 66159 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r66159 | tilghman | 2007-05-25 09:41:27 -0500 - (Fri, 25 May 2007) | 10 lines Merged revisions 66127 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 - May 2007) | 2 lines Issue 9791 - Fix pronunciation of seconds in - Dutch ........ ................ - -2007-05-25 14:37 +0000 [r66158] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_jingle.c, /, configure, configure.ac, - channels/chan_gtalk.c, makeopts.in, res/res_jabber.c: Merged - revisions 66157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) - | 3 lines handle the GNUTLS library properly in the configure - script and build system don't build in OSP support unless we have - found and are allowed to use SSL support ........ - -2007-05-25 13:26 +0000 [r66109-66126] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Minor tweak... drop translation path if one - exists when we get an already signed linear frame in. Chances are - the stream has then switched to signed linear and we no longer - need the path. - - * /, main/slinfactory.c: Merged revisions 66074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66074 | file | 2007-05-24 18:16:58 -0400 (Thu, 24 May 2007) | 2 - lines Fix slinfactory logic when dealing with frames coming in - that may already be in the signed linear format. ........ - -2007-05-24 22:25 +0000 [r66072-66077] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 66076 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) | - 1 line if the string field init fails, clean up the stuff that - was allocated already ........ - - * main/channel.c, /: Merged revisions 66070 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) | - 2 lines Check the result of ast_string_field_init() in - ast_channel_alloc() ........ - -2007-05-24 22:07 +0000 [r66071] Kevin P. Fleming <kpfleming@digium.com> - - * main/aescrypt.c, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, include/asterisk/aes_internal.h - (added), configure.ac, main/aestab.c, include/asterisk/aes.h, - main/aeskey.c, pbx/pbx_dundi.c, channels/chan_iax2.c, - makeopts.in: use the OpenSSL AES implementation if it's available - (unless configured not to) - -2007-05-24 20:55 +0000 [r66031] Jason Parker <jparker@digium.com> - - * /, configure, configure.ac: Merged revisions 66029-66030 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r66029 | qwell | 2007-05-24 15:53:18 -0500 (Thu, 24 May 2007) | 2 - lines Following moving strip to AC_PATH_TOOL, we need to do - something similar for ar. ........ r66030 | qwell | 2007-05-24 - 15:54:16 -0500 (Thu, 24 May 2007) | 2 lines Rebuild configure - script for previous ar fix. ........ - -2007-05-24 20:51 +0000 [r66028] Joshua Colp <jcolp@digium.com> - - * CHANGES, apps/app_voicemail.c: Add ListAllVoicemailUsers manager - command. (issue #8112 reported by Tony Zhao) - -2007-05-24 20:44 +0000 [r65982-66027] Russell Bryant <russell@digium.com> - - * /, configure, configure.ac: Merged revisions 66026 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r66026 | russell | 2007-05-24 15:42:53 -0500 (Thu, 24 - May 2007) | 3 lines Checking for the strip application needs to - be done with AC_PATH_TOOL instead of AC_PATH_PROG to properly - handle cross compilation environments. ........ - - * doc/CODING-GUIDELINES: add a note about using the intenal API for - creating detached threads - - * Makefile, /: Merged revisions 65978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65978 | russell | 2007-05-24 14:05:08 -0500 (Thu, 24 May 2007) | - 3 lines Clear CFLAGS before running make for menuselect. (issue - #9784, reported by ovi, patch by me) ........ - -2007-05-24 19:05 +0000 [r65979] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_gtalk.c: Merged revisions 65965-65967 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) - | 2 lines don't use uninitialized variables ........ r65966 | - kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 - lines don't reference GnuTLS headers and functions unless the - configure script found it ........ r65967 | kpfleming | - 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use - #ifdef instead of #if ........ - -2007-05-24 18:30 +0000 [r65964-65968] Russell Bryant <russell@digium.com> - - * main/pbx.c, include/asterisk/utils.h, channels/chan_zap.c, - channels/chan_sip.c, apps/app_meetme.c, main/utils.c, - channels/chan_iax2.c, main/cdr.c, main/manager.c, - pbx/pbx_spool.c, channels/chan_skinny.c, main/http.c, - channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_rpt.c, - apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: Add - a new API call for creating detached threads. Then, go replace - all of the places in the code where the same block of code for - creating detached threads was replicated. (patch from bbryant) - - * main/rtp.c: Make this build on *my* machine again, and hopefully - not break others. - -2007-05-24 15:35 +0000 [r65906] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, funcs/func_math.c: Merged revisions 65866 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65866 | dhubbard | 2007-05-24 10:08:56 -0500 (Thu, 24 May 2007) - | 1 line merged qwell's func_math patch for issue 9507 ........ - -2007-05-24 15:30 +0000 [r65905] Joshua Colp <jcolp@digium.com> - - * main/manager.c, /: Merged revisions 65902 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65902 | file | 2007-05-24 11:27:23 -0400 (Thu, 24 May 2007) | 2 - lines Add the ability to blacklist certain commands from being - executed using the Command AMI action. (issue #9240 reported by - junky) ........ - -2007-05-24 15:29 +0000 [r65904] Olle Johansson <oej@edvina.net> - - * /, channels/chan_gtalk.c: Merged revisions 65901 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May - 2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump - since the GnuTLS interface did not support multithreading - correctly. ........ - -2007-05-24 15:28 +0000 [r65903] Jason Parker <jparker@digium.com> - - * /, codecs/codec_speex.c, main/translate.c, codecs/codec_ilbc.c, - .cleancount, include/asterisk/translate.h: Merged revisions 65877 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4 - lines Fix handling of zero-length frames when a codec is capable - of native PLC. Issue 9183, patch by Mihai. ........ - -2007-05-24 15:23 +0000 [r65894-65898] Olle Johansson <oej@edvina.net> - - * /, channels/chan_gtalk.c: Merged revisions 65892 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May - 2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by - phsultan. Thanks! ........ - - * /, channels/chan_gtalk.c: Merged revisions 65857 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May - 2007) | 2 lines Issue 7686, fix by phsultan, NAT issues when - calling from gtalk to SIP over nat. ........ - -2007-05-24 15:10 +0000 [r65869] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 65863 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 - lines I like it when the RTP stack compiles myself... ........ - -2007-05-24 15:04 +0000 [r65855] Russell Bryant <russell@digium.com> - - * /, apps/app_festival.c: Merged revisions 65853 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) | - 4 lines Ensure that frames are fully initialized. This will - probably fix getting weird timestamp log messages in logs when - using the Festival app. (issue #9781, patch by me) ........ - -2007-05-24 14:52 +0000 [r65844] Olle Johansson <oej@edvina.net> - - * /, channels/chan_gtalk.c: Merged revisions 65841 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May - 2007) | 2 lines Issue #8536 - Caller ID not set in CDR for jingle - ........ - -2007-05-24 14:50 +0000 [r65843] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: Merged revisions 65842 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | - 5 lines Fix the calculation of the RTT for RTCP. The previous - code would result in oscillating and incorrect data. - Additionally, the RTT would sometimes report negative values due - to incorrect calculations. (issue #9601, patch from davetroy) - ........ - -2007-05-24 14:43 +0000 [r65840] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 65839 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, - 24 May 2007) | 10 lines Merged revisions 65837 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 - lines Allow RFC2833 to be negotiated when an INVITE comes in - without SDP and is not matched to a user or peer. (issue #9546 - reported by mcrawford) ........ ................ - -2007-05-24 14:41 +0000 [r65838] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, res/res_jabber.c: Issue #8409 and - accidentally a fix to chan_sip that wasn't supposed to be there - but is still ok... Sorry. Lack of Tea, really. - -2007-05-24 11:38 +0000 [r65814] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: Yes Virginia, there is a reason why we have - stringfields in the sip_pvt structure... - -2007-05-24 09:51 +0000 [r65769] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65768 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65768 | crichter | 2007-05-24 11:37:32 +0200 - (Do, 24 Mai 2007) | 9 lines Merged revisions 65767 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 - Mai 2007) | 1 line we should only activate the generator in - chan_misdn, when asterisk hask not yet taken the call - (WAITING4DIGS state). Alerting audio will be generated fomr - asterisk for example. ........ ................ - -2007-05-24 03:28 +0000 [r65749] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: - Remove debug variable that was only used - in one place - convert string handling to the ast_str API - - Convert strdup() to ast_strdup() and check the result - Minor - formatting changes - -2007-05-24 03:27 +0000 [r65748] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Oops, should have released this when we - were done with it. - -2007-05-24 02:23 +0000 [r65731] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c: Add SendURL support - -2007-05-23 21:01 +0000 [r65678-65688] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 65685 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) - | 2 lines start the delayed PBX when receive voice or video full - frames as well, and comment this delayed-PBX activity ........ - - * /, channels/chan_sip.c: Merged revisions 65683 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65683 | kpfleming | 2007-05-23 16:51:56 -0400 - (Wed, 23 May 2007) | 10 lines Merged revisions 65682 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 - May 2007) | 2 lines ensure that variables are set on a newly - created channel before we start a PBX on it ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 65679-65680 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) - | 2 lines don't start a PBX on a new incoming IAX2 channel until - we have some sort of response to our ACCEPT (ACK or anything - else) ........ r65680 | kpfleming | 2007-05-23 16:35:50 -0400 - (Wed, 23 May 2007) | 2 lines clear the 'delay PBX' flag when we - are ready to start the PBX ........ - - * /, channels/chan_iax2.c: Merged revisions 65677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65677 | kpfleming | 2007-05-23 16:07:59 -0400 - (Wed, 23 May 2007) | 10 lines Merged revisions 65676 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 - May 2007) | 2 lines if we are going to set variables on a newly - created channel, it should be done *before* we start the PBX on - it ........ ................ - -2007-05-23 17:17 +0000 [r65659] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Don't check for MWI event subscribers - before creating the MWI event in voicemail. MWI events get - cached, so go ahead and always generate them so the cache gets - populated. - -2007-05-23 15:37 +0000 [r65640] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we get the cause code in the REL - -2007-05-23 13:10 +0000 [r65591] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 65589 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65589 | russell | 2007-05-23 08:07:13 -0500 - (Wed, 23 May 2007) | 11 lines Merged revisions 65588 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 - May 2007) | 3 lines Revert revision 62417 as someone reported - problems with it to Mark. This was related to issue #9588. - ........ ................ - -2007-05-23 13:07 +0000 [r65590] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Fix compiling of res_musiconhold under dev - mode. - -2007-05-23 02:55 +0000 [r65573] Russell Bryant <russell@digium.com> - - * main/devicestate.c: Fix a couple minor spelling mistakes. - -2007-05-22 20:26 +0000 [r65542] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/make_version: Merged revisions 65541 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r65541 | kpfleming | 2007-05-22 16:25:41 -0400 (Tue, 22 - May 2007) | 2 lines when building a version string for a - developer branch, include the base branch in the version string - ........ - -2007-05-22 18:52 +0000 [r65502-65505] Russell Bryant <russell@digium.com> - - * main/channel.c, configs/musiconhold.conf.sample, - include/asterisk/channel.h, res/res_musiconhold.c, CHANGES: Add a - new feature for Music on Hold. If you set the "digit" option for - a class in musiconhold.conf, a caller on hold may press this - digit to switch to listening to that music class. This involved - adding a new callback for generators, which allow generators to - get notified of DTMF from the channel they are running on. Then, - a callback was implemented for the music on hold generators. - (patch from bbryant) - - * channels/chan_zap.c, /, apps/app_voicemail.c: Merged revisions - 65501 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | - 3 lines List res_smdi as a dependency for app_voicemail and - chan_zap (Thanks to mnicholson for pointing it out) ........ - -2007-05-22 15:25 +0000 [r65455] BJ Weschke <bweschke@btwtech.com> - - * /, apps/app_followme.c: Merged revisions 65408 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65408 | bweschke | 2007-05-22 10:02:56 -0400 (Tue, 22 May 2007) - | 3 lines Fix a problem with flag recognition. ........ - -2007-05-22 15:08 +0000 [r65451-65454] Joshua Colp <jcolp@digium.com> - - * channels/chan_agent.c: Use ast_strlen_zero where possible. (issue - #9774 reported by eliel) - - * main/cdr.c: Make my compiler happy! Yay! - -2007-05-22 12:58 +0000 [r65376] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Don't overwrite a pointer to the first - channel... that is bad. (issue #9770 reported by tfbu) - -2007-05-22 12:52 +0000 [r65375] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix a couple of spots in the handling of device - states that could lead to a double free. (issue #9772, reported - by Mike Anikienko, fix by me) - -2007-05-22 08:21 +0000 [r65343] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65342 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65342 | crichter | 2007-05-22 10:12:20 +0200 - (Di, 22 Mai 2007) | 9 lines Merged revisions 65328 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 - Mai 2007) | 1 line we stop the tones only when we're in the - pre-call phase, otherwise e.g. when in CONNECTED state we should - not stop tones when we receive an Information Message ........ - ................ - -2007-05-22 02:41 +0000 [r65313] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Fix for 64-bit platform - -2007-05-21 06:56 +0000 [r65298] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: I know we have talked about rewriting app_queue - for Asterisk 1.6, but once I saw this, I couldn't help myself - from changing it. Previously, for *every* device state change, - app_queue would spawn a thread to handle it. Now, the device - state callback just puts the state change in a queue and it gets - handled by a single state change processing thread. - -2007-05-21 02:05 +0000 [r65283] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c: Comment a few more things, and remove an - unnecessary db connection check - -2007-05-20 18:01 +0000 [r65233-65253] Joshua Colp <jcolp@digium.com> - - * /, res/res_agi.c: Merged revisions 65250 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65250 | file | 2007-05-20 13:59:58 -0400 (Sun, 20 May 2007) | 2 - lines res_agi needs to export two symbols (ast_agi_register and - ast_agi_unregister) for usage by others. (issue #9755 reported by - mnicholson) ........ - - * res/res_crypto.c, res/res_musiconhold.c: Music on hold and crypto - no longer need their symbols globally exported. They register the - function pointers upon loading with their respective stubs. - - * main/adsistub.c, main/cryptostub.c: Clean up adsistub file a bit - (just spacing) and change over the crypto sub to use this - build_stub macro strategy. - - * main/Makefile, main/adsistub.c, res/res_adsi.c: Add the adsistub - file to the Asterisk makefile, fix a stub definition, and no - longer make the symbols from res_adsi global since they don't - need to be. - -2007-05-18 22:35 +0000 [r65202-65203] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 65201 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1 - line Ugh. The svnmerge didn't catch the shift from cdr.c to - main/cdr.c, and neither did I. This is the remainder of the 9717 - patch, the fix for the run-away FAIL status for a call ........ - - * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions - 65200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, - 18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 - line This update will fix the situation that occurs as described - by 9717, where when several targets are specified for a dial, if - any one them reports FAIL, the whole call gets FAIL, even though - others were ringing OK. I rearranged the priorities, so that a - new disposition, NULL, is at the lowest level, and the - disposition get init'd to NULL. Then, next up is FAIL, and next - up is BUSY, then NOANSWER, then ANSWERED. All the related set - routines will only do so if the disposition value to be set to is - greater than what's already there. This gives the intended - effect. So, if all the targets are busy, you'd get BUSY for the - call disposition. If all get BUSY, but one, and that one rings is - not answered, you get NOANSWER. If by some freak of nature, the - NULL value doesn't get overridden, then the disp2str routine will - report NOANSWER as before. ........ ................ - -2007-05-18 20:21 +0000 [r65169] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c (added), - configs/cdr_adaptive_odbc.conf.sample (added): Merge - cdr_adaptive_odbc from developer branch - -2007-05-18 18:18 +0000 [r65077-65124] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Related to issue #9235 btw. Merged - revisions 65123 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, - 18 May 2007) | 10 lines Merged revisions 65122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 - lines Not getting an ACK to a 200 OK in the initial invite is - critical to the call. ........ ................ - - * /, channels/chan_sip.c: Merged revisions 65076 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, - 18 May 2007) | 13 lines Merged revisions 65075 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 - lines Issue 9235 - part of the problem, maybe not all. Please - retry with this patch (and no other patch) if you have problems - with hanging SIP channels. Thank you. A special Thank You to - WeBRainstorm that gave me access to his system. ........ - ................ - -2007-05-18 12:43 +0000 [r65006-65040] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged - revisions 65039 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r65039 | crichter | 2007-05-18 14:40:46 +0200 - (Fr, 18 Mai 2007) | 9 lines Merged revisions 65007 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 - Mai 2007) | 1 line fixed a warning regarding Keypad encoding. - encode the IE sending_complete at the right position. ........ - ................ - - * channels/chan_misdn.c, /: Merged revisions 64904 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64904 | crichter | 2007-05-18 10:58:51 +0200 - (Fr, 18 Mai 2007) | 9 lines Merged revisions 64902 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 - Mai 2007) | 1 line we *need* to send a PROCEEDING when - sending_complete is set, even if need_more_infos is requested. - ........ ................ - -2007-05-18 10:41 +0000 [r64973-64975] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 64974 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 - lines Issue 9487 - stop media flows at hangup of call ........ - - * channels/chan_sip.c: Makeup, darling. - -2007-05-18 10:03 +0000 [r64951-64963] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 64515 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64515 | crichter | 2007-05-16 10:44:51 +0200 - (Mi, 16 Mai 2007) | 9 lines Merged revisions 64513 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 - Mai 2007) | 1 line in the case immediate=yes, we directly jump - into the dialplan, where people can use PlayTones to indicate a - Dialtone, so we don't need to to that by ourself. also we should - not do a dialtone_indicate for incoming calls on a TE port in - overlapdialmode. ........ ................ - - * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, - channels/misdn/isdn_lib.c: Merged revisions 63534 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63534 | crichter | 2007-05-09 15:17:10 +0200 - (Mi, 09 Mai 2007) | 17 lines Merged revisions 62945,63402,63519 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | - 1 line when we're in state WAITING4DIGS, we use the asterisk - tone-generator which prods us, so we can't just return -1 in - misdn_write in this case. Added a MISDN_KEYPAD channel variable, - and fixed the sending of keypad. this enables us to modify the - call forward parameters in the switch. ........ r63402 | crichter - | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added - application misdn_check_l2l1 which tries to pull up the L1/L2 on - all ports that have the layers down in a group. It waits then for - a timeout. This helps for scenarios where multiple PMP BRIs are - grouped together, or where a provider has a faulty PTP - Implementation, that looses the L2 after a while. ........ r63519 - | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line - release_chan frees ch, so we should never touch ch after - release_chan, this may cause segfaults. ........ ................ - - * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - Merged revisions 62912 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62912 | crichter | 2007-05-03 16:36:32 +0200 - (Do, 03 Mai 2007) | 17 lines Merged revisions 61357,61770,62885 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | - 1 line some fixes for PMP Hold/Retrieve, it should work now, when - briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 - (Di, 24 Apr 2007) | 1 line added lock for sending messages to - avoid double sending. shuffled some empty_chans after the - cb_event calls, this avoids that a release_complete from a quite - different call releases a fresh created setup by accident. - ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 - Mai 2007) | 1 line fixed the problem that misdn_write did not - return -1 when called with 0 samples in a frame this resultet in - a deadlock in some circumstances, when the call ended because of - a busy extension. added encoding of keypad. ........ - ................ - - * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, - channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged - revisions 59774 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59774 | crichter | 2007-04-03 09:20:27 +0200 - (Di, 03 Apr 2007) | 17 lines Merged revisions 59623-59624,59639 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | - 1 line we can now make 30 channels on a PRI (before we forgot - chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 - (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ - r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | - 1 line added option which allows us to accept incoming SETUP - Messages without automatically sending Proceeding or Setup - Acknowledge, this is useful with some broken switches and if you - want to Release incoming calls without previously having - acknowledged them. The new option is - noautorespond_on_setup=yes|no default is no, so we don't break - the existing behaviour ........ ................ - - * channels/chan_misdn.c, /: Merged revisions 59254 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59254 | crichter | 2007-03-27 17:00:10 +0200 - (Di, 27 Mär 2007) | 9 lines Merged revisions 59252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 - Mär 2007) | 1 line fixed #9355 ........ ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, channels/misdn/isdn_lib.c, - channels/misdn_config.c: Merged revisions 59064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59064 | crichter | 2007-03-20 14:16:06 +0100 - (Di, 20 Mär 2007) | 21 lines Merged revisions - 58849-58850,59062-59063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | - 1 line added method standard_dec for dialing out on groups, to - avoid conflicts, which caused issues with some ISDN providers - ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 - Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | - crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line - avoid sending a disconnect when we already received one. ........ - r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | - 1 line modified a loglevel ........ ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - channels/misdn/isdn_lib.c: Merged revisions 58825-58826 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58825 | crichter | 2007-03-12 13:43:24 +0100 - (Mo, 12 Mär 2007) | 1 line added UU transceiving and corect - handling for rdnis ................ r58826 | crichter | - 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines Merged - revisions 57034,57523,57753,58558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | - 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com - bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 - 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ - r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | - 1 line fixed another place where the out_cause was hardcoded to - 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 - Mar 2007) | 1 line we can free channel 31 as well, since we can - occupy it ........ ................ - -2007-05-18 09:10 +0000 [r64903-64921] Olle Johansson <oej@edvina.net> - - * include/asterisk/adsi.h, main/adsistub.c (added), res/res_adsi.c, - apps/app_voicemail.c: Issue #5930 - Remove dependencies on - res_adsi.so - clwade A big THANK YOU to clwade for this patch. - Minor modifications by me. - - * channels/chan_sip.c: Another fix for the support for recordings - controlled by INFO-packets We still lack a setting to - enable/disable this per peer - -2007-05-18 02:55 +0000 [r64869-64870] Russell Bryant <russell@digium.com> - - * CHANGES: Add ENUMQUERY and ENUMRESULT to the CHANGES file. - - * /, apps/app_queue.c: Merged revisions 64868 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) | - 5 lines Fix a small bug I noticed while working on something - else. app_queue did not unregister its device state monitoring - callback in unload_module(). So, this would make Asterisk crash - on the first device state change after you unload the module. - ........ - -2007-05-17 21:20 +0000 [r64821] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 64820 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64820 | tilghman | 2007-05-17 16:19:34 -0500 - (Thu, 17 May 2007) | 10 lines Merged revisions 64819 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 - May 2007) | 2 lines How is it that we never caught that this is - returning the opposite of our documentation, until now? ........ - ................ - -2007-05-17 17:12 +0000 [r64786] Russell Bryant <russell@digium.com> - - * main/manager.c, configs/manager.conf.sample: Add an option that - lets you only allow one connection at a time for each manager - user. (issue #8664, reported and original patch by ssokol, patch - updated by bkruse, and further updated by me) - -2007-05-17 16:54 +0000 [r64762] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 64761 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64761 | qwell | 2007-05-17 11:53:27 -0500 (Thu, - 17 May 2007) | 12 lines Merged revisions 64758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 - lines If we have a negative current message, we shouldn't go back - even further... Issue 9727. ........ ................ - -2007-05-17 16:53 +0000 [r64757-64760] Russell Bryant <russell@digium.com> - - * /, contrib/scripts/astxs (removed): Merged revisions 64759 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64759 | russell | 2007-05-17 11:52:53 -0500 (Thu, 17 May 2007) | - 3 lines Remove script that is no longer functional since the - build system was redone. (issue #9340, reported by junky) - ........ - - * apps/app_dial.c, /: Merged revisions 64756 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) | - 3 lines Increase the size of a buffer to support longer dial - strings for channels. (issue #9291, reported and fix suggested by - meni) ........ - -2007-05-17 16:11 +0000 [r64721-64755] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 64754 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 - lines Even more direct RTP setup fixes! Don't allow a codec that - isn't supported to creep into the SDP of either side. (issue - #9446 reported by marcelbarbulescu) ........ - - * /, apps/app_voicemail.c: Merged revisions 64720 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64720 | file | 2007-05-17 09:48:44 -0400 (Thu, 17 May 2007) | 2 - lines Fix authuser support. (issue #9740 reported by - xmarksthespot) ........ - -2007-05-17 06:14 +0000 [r64657-64687] Russell Bryant <russell@digium.com> - - * README, /: Merged revisions 64686 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64686 | russell | 2007-05-17 01:13:53 -0500 (Thu, 17 May 2007) | - 3 lines Update the main README to reflect the new build process - for 1.4 and above. (issue #9725, patch by eliel) ........ - - * main/app.c: Ignore this ... playing with jira (AST-1) - -2007-05-16 11:01 +0000 [r64494-64611] Olle Johansson <oej@edvina.net> - - * /: Blocking patch - - * /, channels/chan_sip.c: Below patches with some re-structuring - for trunk --- Merged revisions 64602 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 - lines Issue #9681 - Handle www-auth on BYE ........ - - * /, channels/chan_sip.c: Merged revisions 64578 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 - lines Final part of issue #9483 - fixing transfer() of sip calls - in the dial plan (twilson) ........ - - * /: Blocking patch that was already committed to trunk - - * /, channels/chan_sip.c: Merged revisions 64543 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, - 16 May 2007) | 10 lines Merged revisions 64535 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 - lines Support SIP uri's starting with SIP: and sip: (reported by - Tony Mountfield on the mailing list. Thanks!) ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 64516 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, - 16 May 2007) | 17 lines Merged following patch with a lot of - changes for 1.4 ------ Merged revisions 64514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 - lines Issue #9726 - rlister - Better logging for ACL denials - While at it, also added better logging and handling of peers that - are not supposed to register. My patch, stole the issue report - from Russell. My apologies, Russell :-) ........ ................ - - * channels/chan_sip.c: Issue #9304 - Update help text to match - functionality. Patch by kshumard with changes by oej - - * channels/chan_sip.c, configs/sip.conf.sample: Issue #6789 - - Marquis - Add option to support regexten removal when host - becomes unreachable - - * main/event.c: This file really needs more documentation... When - we implement new API's - please include a small general overview - in Doxygen - - * main/dial.c: Small doxygen updates - -2007-05-15 23:05 +0000 [r64469-64480] Russell Bryant <russell@digium.com> - - * funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: Add two - new dialplan functions: ENUMQUERY and ENUMRESULT. These functions - allow you to initiate an ENUM query using ENUMQUERY, and then - access the details of all of the results using ENUMRESULT. - Previously, if you wanted to access multiple results, Asterisk - would have to do a new DNS lookup every time. (patch by bbryant) - - * pbx/pbx_dundi.c: Make sure that DUNDIRESULT is given an ID. - -2007-05-15 20:45 +0000 [r64455] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: XXX-XXX-XXX - appears to be the standard ANSI pointcode format - -2007-05-15 19:57 +0000 [r64427] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 64426 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64426 | russell | 2007-05-15 14:52:18 -0500 (Tue, 15 May 2007) | - 3 lines Properly fix a problem that occurs when you set - PARKINGEXTEN to an exten where a call is already parked. (issue - #9723, patch by me) ........ - -2007-05-14 23:43 +0000 [r64399] Kevin P. Fleming <kpfleming@digium.com> - - * /: this does not belong here - -2007-05-14 22:25 +0000 [r64384] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Only print the SS7 UP once. Not every time - we get the test messages on the line. - -2007-05-14 21:51 +0000 [r64355] Jason Parker <jparker@digium.com> - - * main/Makefile: With libmmime.a as a .PHONY target, asterisk gets - rebuilt every time, but without proper ASTCFLAGS. This caused a - problem with the buildinfo.o file not being able to find - asterisk/build.h This was affecting DESTDIR, but I *think* that - if asterisk had never been installed before, it would've failed - also. - -2007-05-14 21:17 +0000 [r64354] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 64353 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64353 | russell | 2007-05-14 16:16:39 -0500 (Mon, 14 May 2007) | - 4 lines When someone requests a specific parking space using the - PARKINGEXTEN variable, ensure that no other caller is already - there. (issue #9723, reported by mdu113, patch by me) ........ - -2007-05-14 19:35 +0000 [r64323-64325] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 64324 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 - lines Change -2 to XMIT_ERROR to clarify a bit more ........ - - * /: Blocking patch already committed to trunk - -2007-05-14 19:21 +0000 [r64322] Russell Bryant <russell@digium.com> - - * /, channels/chan_alsa.c: Merged revisions 64306 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | - 3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it. - An unknown indication will trigger an error and cause sounds to - stop, which in this case, is ringing. ........ - -2007-05-14 18:49 +0000 [r64274-64279] Joshua Colp <jcolp@digium.com> - - * /, codecs/codec_speex.c: Merged revisions 64278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64278 | file | 2007-05-14 14:48:33 -0400 (Mon, 14 May 2007) | 2 - lines Properly set datalen field when doing PLC in codec_speex. - (issue #9722 reported by mihai) ........ - - * /, main/devicestate.c: Merged revisions 64276 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r64276 | file | 2007-05-14 14:36:34 -0400 (Mon, - 14 May 2007) | 10 lines Merged revisions 64275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 - lines Only perform stripping of - strings from the channel name - for Zap channels. Anywhere else we might remove a legitimate part - of a device name. (issue #9668 reported by stevedavies) ........ - ................ - - * channels/chan_sip.c: If no port is specified in the outboundproxy - setting then use the standard SIP port. (issue #9665 reported by - tootai) - -2007-05-14 18:14 +0000 [r64243-64273] Jason Parker <jparker@digium.com> - - * configs/queues.conf.sample: oops - silly typo there - - * configs/queues.conf.sample, apps/app_queue.c: Don't allow - rounding seconds to weird values that may cause "unexpected" - results. Issue 9514. - - * apps/app_queue.c: Add 'c' option to app_queue which allows for - continuing in the dialplan if the callee hangs up. Issue 9284, - patch by lyl, modified a little bit by me (I felt 'continue' was - better than 'keepalive') - -2007-05-14 17:25 +0000 [r64242] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 64240 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2 - lines Fix scenario where if a phone that simply called Echo() put - itself on hold it could never get off hold. ........ - -2007-05-14 16:08 +0000 [r64225-64226] Russell Bryant <russell@digium.com> - - * configure: Regenerate configure script after last change to - acinclude.m4 - - * acinclude.m4: Remove an extra space from the macro that checks - for C defines. (issue #9715, tzafrir) - -2007-05-14 14:13 +0000 [r64208] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, channels/chan_local.c, /: Merged - revisions 64193 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 - line As per 9570, worrisome CDR warnings have been removed, that - are either not helpful, or not relevant. ........ - -2007-05-14 10:40 +0000 [r64142-64158] Olle Johansson <oej@edvina.net> - - * main/channel.c, /: Merged revisions 64157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2 - lines Add hangupcause when we lack codecs for transcoding - ........ - - * channels/chan_sip.c: Improve handling network errors on - transmission to hosts that don't reply or are unreachable With - this code, the call will fail as soon as we get a network error. - This may happen on first xmit or a later one, so the retransmit - code handles this too. - -2007-05-12 22:28 +0000 [r64087-64115] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 64114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 - lines This concludes my final adventure with bitmasks and the - onhold flag. Would anyone care for some peanuts? ........ - - * /, channels/chan_sip.c: Merged revisions 64086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 - lines Tweak hold flags some more. They can be of three states - when active: active, inactive, one direction. ........ - -2007-05-12 19:38 +0000 [r64072] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_enum.c: Issue 9716 - doc/enum.txt no longer exists in - trunk - -2007-05-12 16:33 +0000 [r64045] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 64044 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 - lines Ensure the onhold flag is set no matter what when being put - on hold. ........ - -2007-05-11 22:52 +0000 [r63967-64030] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Add/fix - support for Redial, Speeddial, and Messages buttons. Combined - effort by DEA and mvanbaak. - - * main/asterisk.c: oops.. Fix the logic of the last commit. - - * Makefile, main/asterisk.c: Better fallback method for - autosystemname. Issue 9713, patch by Juggie with minor mods by - me. - - * main/manager.c, /: Merged revisions 63982 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7 - lines Hide manager password from "manager show user foo". I - realize that there are other ways to get this, but we really - don't need to just show it in plain text so easily. Issue 9273, - patch by junky ........ - - * Makefile, main/asterisk.c: Add autosystemname setting to - asterisk.conf When enabled, it will set the systemname to be the - hostname of the system Issue 9713, patch by Juggie - slightly - modified by me, to "failover" to localhost - -2007-05-11 18:31 +0000 [r63946] Russell Bryant <russell@digium.com> - - * doc/qos.tex: Fix some syntax errors. - -2007-05-11 16:37 +0000 [r63906] Tilghman Lesher <tlesher@digium.com> - - * Makefile, /, contrib/scripts/safe_asterisk: Merged revisions - 63905 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63905 | tilghman | 2007-05-11 11:35:51 -0500 - (Fri, 11 May 2007) | 10 lines Merged revisions 63903 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 - May 2007) | 2 lines Issue 9121 - fixups for safe_asterisk script - ........ ................ - -2007-05-11 16:21 +0000 [r63901-63902] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 63886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) | - 6 lines When MD5 authentication is not possible because there is - no challenge present, either because the Challenge action was - never issued, or some other reason, give a proper error message - and return an error instead of claiming that the user wasn't - found. (reported by jsmith on IRC) ........ - - * res/res_agi.c: Add gender support for AGI SAY NUMBER. (issue - #9537, patch by chappell) - -2007-05-11 15:48 +0000 [r63873] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 63872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63872 | file | 2007-05-11 11:43:14 -0400 (Fri, 11 May 2007) | 2 - lines Make the PARKINGEXTEN feature of parking actually work. - (issue #9708 reported by mdu113) ........ - -2007-05-10 23:16 +0000 [r63832] Jason Parker <jparker@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 63830 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, - 10 May 2007) | 12 lines Merged revisions 63828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 - lines Fix an issue with trying to kill a thread before it gets - created. Issue 9709, patch by nic_bellamy. ........ - ................ - -2007-05-10 22:25 +0000 [r63805] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 63804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63804 | russell | 2007-05-10 17:23:42 -0500 (Thu, 10 May 2007) | - 4 lines Strip terminal escape sequences from CLI command output - that is going to be sent out over the manager interface. (issue - #9659, reported by pari, fixed by me) ........ - -2007-05-10 21:25 +0000 [r63786] Doug Bailey <dbailey@digium.com> - - * main/callerid.c: Added check for negative offset in cid spill to - prevent infinite loops - -2007-05-10 20:51 +0000 [r63730-63751] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 63749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, - 10 May 2007) | 12 lines Merged revisions 63748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 - lines Do not allocate SIP pvt's for PEERs we can not reach. This - was seen as a lot of dialogs being created then immediately - destroyed at reload/restart of the SIP channel. ........ - ................ - - * apps/app_minivm.c: Fixing reload. Thanks to Mats Karlsson! - -2007-05-09 19:24 +0000 [r63699] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 63698 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2 - lines Use the DTMF frame on the channel when returning a DTMF - frame from AST_FRAME_NULL or AST_FRAME_VOICE. ........ - -2007-05-09 19:21 +0000 [r63697] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 63612 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | - 5 lines Modify ast_senddigit_begin() to use the same assumptions - used elsewhere in the code in that if a channel does not have a - send_digit_begin() callback, it only cares about DTMF END events. - (pointed out by Michael Neuhauser on the asterisk-dev list) - ........ - -2007-05-09 17:35 +0000 [r63655] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Merged revisions 63654 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, - 09 May 2007) | 10 lines Merged revisions 63653 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 - lines Make sure we only create a DSP if it's requested on - SUB_REAL ........ ................ - -2007-05-09 16:56 +0000 [r63613] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 63611 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, - 09 May 2007) | 10 lines Merged revisions 63610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 - lines Properly handle hints that point to multiple devices in - chan_sip. Why chan_sip is even doing this I have no idea but I - would rather not go into a rant. (issue #9536 reported by - rlister) ........ ................ - -2007-05-09 16:44 +0000 [r63609] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 63608 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | - 5 lines Only call ast_senddigit_begin() in ast_senddigit() if the - channel has a send_digit_begin() callback. Checking the - END_DTMF_ONLY flag was the wrong thing to do, because that flag - indicates that a *bridged* channel only wants DTMF END events - coming from this channel. ........ - -2007-05-09 14:52 +0000 [r63567] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_directory.c: Merged revisions 63566 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63566 | tilghman | 2007-05-09 09:50:33 -0500 - (Wed, 09 May 2007) | 10 lines Merged revisions 63565 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 - May 2007) | 2 lines Replicate fix from 51158 (app_voicemail) to - app_directory (Issue 9224) ........ ................ - -2007-05-09 13:24 +0000 [r63536] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 63535 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63535 | russell | 2007-05-09 08:24:03 -0500 (Wed, 09 May 2007) | - 6 lines I have seen multiple people post questions trying to - figure out what the message "The configure script must be - executed before running 'make'" means. So, add another like that - says to specifically run ./configure. If this isn't obvious - enough, then they should be using something like AsteriskNOW and - not installing from source. ........ - -2007-05-09 13:07 +0000 [r63533] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 63532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 - lines Don't retransmit 200 OK's on ignore status. (Reported on - asterisk-users) ........ - -2007-05-08 22:40 +0000 [r63479] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_macro.c: Merged revisions 63478 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63478 | tilghman | 2007-05-08 17:38:02 -0500 - (Tue, 08 May 2007) | 10 lines Merged revisions 63477 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 - May 2007) | 2 lines Issue 9602 - segfault in app_macro ........ - ................ - -2007-05-08 16:54 +0000 [r63404-63449] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 63448 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63448 | russell | 2007-05-08 11:53:09 -0500 (Tue, 08 May 2007) | - 4 lines I mixed up the use of the find_feature() function, so I - renamed it find_dynamic_feature, and changed the code to use the - correct lock when using it. ........ - - * channels/chan_sip.c, res/res_features.c, - include/asterisk/features.h: I noted this on the dev list but got - no response, so I just did it myself. Lock the call features when - being used in chan_sip. - - * /, res/res_features.c: Merged revisions 63445 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63445 | russell | 2007-05-08 11:30:43 -0500 (Tue, 08 May 2007) | - 2 lines Use a read/write lock when accessing the built-in - features. ........ - - * contrib/scripts/realtime_pgsql.sql (added), /, - contrib/realtime_pgsql.sql (removed): Merged revisions 63403 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63403 | russell | 2007-05-08 10:10:37 -0500 (Tue, 08 May 2007) | - 3 lines Move realtime_pgsql.sql to contrib/scripts to be with the - rest of the sql examples. (issue #9676, suretec) ........ - -2007-05-08 06:26 +0000 [r63361] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 63360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63360 | tilghman | 2007-05-08 01:22:37 -0500 - (Tue, 08 May 2007) | 10 lines Merged revisions 63359 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 - May 2007) | 2 lines Issue 9527 - upon entering a folder, no - message is selected (curmsg == -1), so deleting causes memory - corruption (beyond bounds) ........ ................ - -2007-05-07 22:32 +0000 [r63319-63330] Russell Bryant <russell@digium.com> - - * /, contrib/realtime_pgsql.sql (added), - configs/res_pgsql.conf.sample (added): Merged revisions 63329 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) | - 3 lines Add a sample configuration file and example tables for - use with res_config_pgsql. (issue #9676, suretec) ........ - - * apps/app_meetme.c: Make a minor tweak to admin_exec() - don't - lock the conference list until it is actually necessary. - - * apps/app_meetme.c, CHANGES: Add a new application, - MeetMeChannelAdmin, which is similar to MeetMeAdmin, except it - lets you operate on a channel by name instead of conference - member number. It is very useful in combination with the 'X' - option to ChanSpy. (issue #9671, patch by mnicholson, with some - small modifications by me) - -2007-05-07 21:47 +0000 [r63284-63287] Joshua Colp <jcolp@digium.com> - - * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged - revisions 63286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, - 07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 - lines Properly handle what happens during a masquerade in - relation to group counting. (issue #9657 reported by ramonpeek) - ........ ................ - -2007-05-07 20:07 +0000 [r63228-63255] Olle Johansson <oej@edvina.net> - - * /, main/config.c: Merged revisions 63254 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63254 | oej | 2007-05-07 22:05:15 +0200 (Mon, 07 May 2007) | 2 - lines Don't remove configuration from memory just because one - section failed. ........ - - * include/asterisk/module.h, main/loader.c: Constifications - - * channels/chan_jingle.c, res/res_jabber.c: Adding external - referenses for doxygen See - http://www.asterisk.org/doxygen/trunk/extref.html - - * channels/chan_misdn.c: Adding external reference - - * channels/chan_misdn.c: Doxyfication... There's a shortage of - comments in this file... - -2007-05-06 20:09 +0000 [r63182] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Lock iax2 pvt structure when passing off to - the AMI function, and make sure it exists. (issue #9674 reported - by arabe) - -2007-05-06 13:11 +0000 [r63168] Olle Johansson <oej@edvina.net> - - * /, main/file.c: Merged revisions 63152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63152 | oej | 2007-05-06 14:28:38 +0200 (Sun, 06 May 2007) | 2 - lines Stop the video stream when you stop playback of all streams - for a call ........ - -2007-05-05 08:05 +0000 [r63136] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Adding some missing spaces - Correcting - error messages - Disabling code that doesn't do anything - Making - sure we always respond to this request, happily - -2007-05-04 20:11 +0000 [r63105] Pari Nannapaneni <paripurnachand@digium.com> - - * /, configs/manager.conf.sample: Merged revisions 63047 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1 - line explanation for httptimeout in manager.conf ........ - -2007-05-04 20:06 +0000 [r63104] Jason Parker <jparker@digium.com> - - * /, res/res_jabber.c: Merged revisions 63099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r63099 | qwell | 2007-05-04 15:03:49 -0500 (Fri, 04 May 2007) | 4 - lines Fix a crash when checking version attribute in an incoming - XML caps element. Issue 9667, patch by phsultan. ........ - -2007-05-04 19:48 +0000 [r63089] Russell Bryant <russell@digium.com> - - * main/manager.c: Convert spaces to tabs for indentation. - -2007-05-04 18:47 +0000 [r63046-63076] Steve Murphy <murf@digium.com> - - * res/res_features.c: According to my testing, it's better if the - ast_find_call_feature func ran this way instead, as far as the - snom record button is concerned - - * doc/CODING-GUIDELINES, channels/chan_sip.c, res/res_features.c, - include/asterisk/features.h: a small upgrade to the coding - standard, and an update to the code that triggered the upgrade. - - * channels/chan_sip.c, res/res_features.c, UPGRADE.txt, - include/asterisk/features.h: Added a small bit of code to support - the SNOM 360's Record button. Made the find_feature func in - res_features.c public, so I could use it to find the automon dial - sequence as configured by the user. When the INFO packet has a - Record: header with on/off, the sequence is sent as consecutive - DTMF frames on the phone's channel, triggering the automon - functionality. The user has to configure the automon in - features.conf, and set up his dialplan accordingly. - -2007-05-04 13:56 +0000 [r63030-63032] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, channels/chan_iax2.c: Add the new - ChannelUpdate event to inform manager clients about the PVT ID - and some other channel driver data that is needed to follow the - call through the PBX. - - * main/manager.c: Add "CoreStatus" - from the moremanager branch. - This can be extended with more information, ideas and patches are - welcome, as usual :-) - - * include/asterisk.h, main/manager.c, include/asterisk/manager.h, - include/asterisk/options.h: - Add manager command CoreSettings - - Add missing option to options.h - Add missing variables to - asterisk.h - Move manager version to manager.h include file - -2007-05-03 16:45 +0000 [r62990] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 62989 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, - 03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 - lines When a peer is seeded or built tell the devicestate core to - update it's status. This is easier then having chan_sip load - before pbx_config. (issue #9658 reported by dlynes) ........ - ................ - -2007-05-03 16:43 +0000 [r62988] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/loader.c: Merged revisions 62986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007) - | 2 lines improve loader a bit, by avoiding trying to initialize - embedded modules twice and avoiding trying to load modules from - disk when they have been loaded already during the 'preload' pass - (reported by blitzrage on IRC, patch by me) ........ - -2007-05-03 15:23 +0000 [r62943] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 62942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | - 17 lines Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, - 2007, TM, Patent Pending). This set of changes came from a - debugging session I had with Dwayne Hubbard. When he called into - his home FXO, ran the Echo application, and pressed a digit, the - digit would be echoed back and would never end. This is fixed, - along with a couple other little improvements. * When chan_zap is - in the middle of playing a digit to a channel, it feeds back null - frames, not voice frames. So, I have modified ast_read to check - the timing on emulated DTMF when it receives null frames, in - addition to where it was doing this on voice frames. * Make a - tweak to setting the duration on emulated DTMF digits. If there - was no duration specified, it set it to be the minimum, instead - of the default. * Instead of timing the emulated digits off of - the number of samples in audio frames that pass through, just use - time values. Now there is no code in this section that assumes - 8kHz audio. ........ - -2007-05-03 14:44 +0000 [r62911-62914] Steve Murphy <murf@digium.com> - - * /: blocking 62913 (1.4) from trunk, as it's already done here - - * /, pbx/ael/ael.tab.c, pbx/ael/ael.y, - pbx/ael/ael-test/ref.ael-test20 (added), pbx/ael/ael.tab.h, - pbx/ael/ael-test/ael-test20/extensions.ael (added), - pbx/ael/ael-test/ael-test20 (added): Merged revisions 62883 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62883 | murf | 2007-05-03 07:54:56 -0600 (Thu, 03 May 2007) | 1 - line These mods fix bug 9623, where an '@' in the eswitch - contents causes a syntax error. I also updated the regressions. - ........ - -2007-05-03 00:25 +0000 [r62824-62843] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 62842 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62842 | kpfleming | 2007-05-02 20:23:37 -0400 - (Wed, 02 May 2007) | 10 lines Merged revisions 62841 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 - May 2007) | 2 lines doh... initializing the pointer variable will - work just a bit better ........ ................ - - * main/minimime: ignore the archive we build in this directory - - * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged - revisions 62797,62807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62797 | kpfleming | 2007-05-02 19:57:23 -0400 - (Wed, 02 May 2007) | 7 lines improve static Realtime config - loading from PostgreSQL: don't request sorting on fields that are - pointless to sort on use ast_build_string() instead of snprintf() - don't request the list of fieldnames that resulted from the query - when we both knew what they were before we ran the query _AND_ we - aren't going to do anything with them anyway (patch by me, - inspired by blitzrage's bug report about res_config_odbc) - ................ r62807 | kpfleming | 2007-05-02 20:02:57 -0400 - (Wed, 02 May 2007) | 15 lines Merged revisions 62796 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 - May 2007) | 7 lines increase reliability and efficiency of static - Realtime config loading via ODBC: don't request fields we aren't - going to use don't request sorting on fields that are pointless - to sort on explicitly request the fields we want, because we - can't expect the database to always return them in the order they - were created (reported by blitzrage in person (!), patch by me) - ........ ................ - -2007-05-02 23:50 +0000 [r62791-62795] Russell Bryant <russell@digium.com> - - * CHANGES: Fix some bad grammar. - - * apps/app_meetme.c, CHANGES: When a conference is created, the - UNIQUEID of the channel that caused it to be created will now be - stored. Then, every channel that joins the conference will have - the MEETMEUNIQUEID channel variable set with this ID. This can be - used to relate callers that come and go from long standing - conferences. (issue #7295, patch by softins) - - * CHANGES: Note Hungarian language support in CHANGES - - * main/say.c, configs/say.conf.sample: Add Hungarian language - support to say.c and say.conf. (issue #7077, patch by adomjan) - - * main/channel.c, /: Merged revisions 62789 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | - 20 lines Merge changes from team/russell/inband_dtmf ... Fix some - issues related to generating inband DTMF. There are two changes - here: 1) The list of DTMF tones in the senddigit_begin() function - explicitly specified 100ms of the tone followed by 100ms of - silence. This really broke things with the way that Asterisk now - wants complete control over when the digit begins and ends. So, - regardless of what Asterisk really wanted to do, this was going - to play out the tone at the length it wanted to. This caused - various problems like DTMF translation to inband to be extremely - unreliable. The list of tones has been changed so that the - correct DTMF tone is played indefinitely until Asterisk tells it - to stop. 2) ast_write() had to be modified to let a DTMF_END - frame get processed even when a generator is present. This is how - the tone will finally get stopped. (issues #8944, #9250, #9348, - maybe others. Thanks to mdu113 from #8944 for the testing and - feedback!) ........ - -2007-05-02 20:57 +0000 [r62741] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, /: Merged revisions 62738 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62738 | murf | 2007-05-02 14:46:07 -0600 (Wed, - 02 May 2007) | 9 lines Merged revisions 62737 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1 - line Some tweaks to satisfy CDR bug 8796, where being in 'h' - extension louses up the dst field ........ ................ - -2007-05-02 17:49 +0000 [r62693] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 62692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62692 | tilghman | 2007-05-02 12:43:48 -0500 - (Wed, 02 May 2007) | 12 lines Merged revisions 62691 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 - May 2007) | 4 lines Issue 9638 - if a text frame is sent with no - terminating NULL through a bridged IAX connection, the remote end - will receive garbage characters tacked onto the end. ........ - ................ - -2007-05-02 17:24 +0000 [r62690] Steve Murphy <murf@digium.com> - - * main/channel.c, main/pbx.c, channels/chan_zap.c, /, - cdr/cdr_radius.c: Merged revisions 62689 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 - line a)In chan_zap, set the clid, src fields in channel_alloc - call. b)in the channel_alloc func, set the cid_num and name - fields from the arglist[blush]. c) don't update the channel app & - app data fields if you are in the 'h' extension. d)the - load_module func in cdr_radius needs to return DECLINE, SUCCESS. - ........ - -2007-05-02 15:46 +0000 [r62671-62673] Russell Bryant <russell@digium.com> - - * channels/chan_local.c, CHANGES: Update the device state - functionality of chan_local such that it will return NOT_INUSE or - INUSE when Local channels are in use as opposed to just UNKNOWN. - It will still return INVALID if the extension doesn't exist at - all. (issue #8048, patch from tim_ringenbach) - - * CHANGES: Add the new options for attended transfer to the CHANGES - file. - - * doc/ip-tos.tex (removed), doc/qos.tex (added): For some reason - when I merged 802.1p support, the new documentation file was not - properly added. Thanks to IgorG for pointing it out! :) - -2007-05-02 12:12 +0000 [r62609-62656] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add a small message that we're doing - something. On my systems, there's a long dead period with a - non-responsive CLI after I issue "load chan_sip.so" - - * channels/chan_sip.c: More username body parts to fix... If - working, this needs to be backported to 1.2, 1.4. But first, some - serious SIP testing :-) - - * channels/chan_sip.c: Handle - sip:username;parameter=12345@example.com;parameter=1234 URI's - properly - - * /, channels/chan_sip.c: Merged revisions 62624 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 - lines Don't unlock a channel that we already know does not exist - (propably isue 8228) ........ - - * CREDITS: Updating CREDITS - -2007-05-01 22:24 +0000 [r62549-62593] Russell Bryant <russell@digium.com> - - * res/res_features.c, configs/features.conf.sample: In addition to - making it so attended transfers don't fail unnecessarily, add - some new options to control what happens when you hangup on an - attended transfer before the target extension answers the - transferred channel. You can now have it send the transferee back - to the transferer. (issue #8413, patch from sergee with very - minor modifications by me) - - * /, res/res_features.c: Merged revisions 62548 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62548 | russell | 2007-05-01 16:57:10 -0500 - (Tue, 01 May 2007) | 12 lines Merged revisions 62547 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 - May 2007) | 4 lines Remove an unnecessary check that makes it so - if you hang up after doing an attended transfer before the target - extension answers the channel, the transfer is not successful. - (issue #9338, patch by svanlund) ........ ................ - -2007-05-01 21:41 +0000 [r62546] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 62545 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007) - | 2 lines Bug 9590 - Memory leaks around find_user() (found by - rayjay, different fixes by me) ........ - -2007-05-01 16:27 +0000 [r62415-62498] Russell Bryant <russell@digium.com> - - * /, configs/indications.conf.sample: Merged revisions 62497 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62497 | russell | 2007-05-01 11:26:48 -0500 - (Tue, 01 May 2007) | 11 lines Merged revisions 62496 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 - May 2007) | 3 lines Add indications.conf information for the - Philippines. (issue #9525, reported and patched by loloski) - ........ ................ - - * CHANGES: Add a note to CHANGES about the new support for 802.1p. - Thanks IgorG! - - * CHANGES, apps/app_queue.c, doc/queuelog.tex: This patch adds - additional information to the EXITWITHKEY and EXITWITHTIMEOUT - entries in the queue log. (issue #7561, reported and originally - patched by fkasumovic, patch slightly modified and updated to - trunk by me) - - * include/asterisk/acl.h, main/udptl.c, channels/chan_sip.c, - include/asterisk/rtp.h, main/acl.c, include/asterisk/netsock.h, - channels/iax2-provision.c, channels/chan_iax2.c, main/rtp.c, - main/netsock.c, configs/h323.conf.sample, - configs/iax.conf.sample, configs/mgcp.conf.sample, - configs/iaxprov.conf.sample, channels/chan_h323.c, - pbx/pbx_dundi.c, include/asterisk/udptl.h, - configs/sip.conf.sample, doc/asterisk.tex, channels/chan_mgcp.c: - Add support for setting the CoS for VLAN traffic (802.1p) in - Linux. The file doc/qos.tex has been updated to document the new - functionality. (issue #9540, patch submitted by IgorG) - - * channels/chan_zap.c, /: Merged revisions 62419 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62419 | russell | 2007-04-30 10:58:28 -0500 - (Mon, 30 Apr 2007) | 12 lines Merged revisions 62417 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 - Apr 2007) | 4 lines This patch fixes an issue where depending on - the cause code, when the network sends a PRI disconnect, the call - may not be properly hung up. (issue #9588, reported and patched - by softins) ........ ................ - - * channels/chan_sip.c: Don't crash when invalid arguments are - provided to the CHANNEL() function for a SIP channel. (issue - #9619, reported by jtodd, original patch by Corydon76, committed - patch slightly modified by me) - - * include/asterisk/http.h, /, main/http.c: Merged revisions 62414 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | - 4 lines When serving dynamic content, include a Cache-Control - header to instruct the browsers to not store the resulting - content. (issue #9621, reported by Pari, patch by me) ........ - -2007-04-30 14:56 +0000 [r62372] Jason Parker <jparker@digium.com> - - * configs/iax.conf.sample, /: Merged revisions 62371 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr - 2007) | 2 lines Remove unused (and potentially confusing) - jitterbuffer options from sample config. ........ - -2007-04-30 14:37 +0000 [r62370] Joshua Colp <jcolp@digium.com> - - * /, main/asterisk.c: Merged revisions 62369 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62369 | file | 2007-04-30 11:36:11 -0300 (Mon, - 30 Apr 2007) | 10 lines Merged revisions 62368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 - lines Update copyright notice. It's now the year 2007! ........ - ................ - -2007-04-29 05:51 +0000 [r62219-62332] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 62331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | - 3 lines Fix a bug that made the "language" setting in zapata.conf - not functional. (issue #9626, reported and fixed by sergee) - ........ - - * CHANGES: note MeetMe change in CHANGES - - * apps/app_meetme.c: Enable the functionality of the 'o' option to - "optimize talker" by default. - - * channels/iax2.h: Reformat some of iax2.h and convert comments to - doxygen format - - * include/asterisk.h, channels/chan_zap.c, channels/chan_sip.c, - main/Makefile, res/res_eventtest.c (added), - configs/voicemail.conf.sample, UPGRADE.txt, CHANGES, - channels/chan_iax2.c, main/dial.c, include/asterisk/event.h - (added), include/asterisk/event_defs.h (added), main/event.c - (added), configs/sip.conf.sample, main/asterisk.c, - channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from - team/russell/events This set of changes introduces a new generic - event API for use within Asterisk. I am still working on a way - for events to be shared between servers, but this part is ready - and can already be used inside of Asterisk. This set of changes - introduces the first use of the API, as well. I have restructured - the way that MWI (message waiting indication) is handled. It is - now event based instead of polling based. For example, if there - are a bunch of SIP phones subscribed to mailboxes, then chan_sip - will not have to constantly poll the mailboxes for changes. - app_voicemail will generate events when changes occur. See - UPGRADE.txt and CHANGES for some more information on the effects - of these changes from the user perspective. For developer - information, see the text in include/asterisk/event.h. As always, - additional feedback is welcome on the asterisk-dev mailing list. - - * doc/ast_appdocs.tex, doc/dundi.tex: Update the DUNDi section of - the documentation with example usage of DUNDIQUERY and - DUNDIRESULT. Also, update the automatically generated application - docs. - - * pbx/pbx_dundi.c, CHANGES: Merge changes from - team/russell/dundi_results This introduces two new dialplan - functions: DUNDIQUERY and DUNDIRESULT. DUNDIQUERY lets you - intitiate a DUNDi query from the dialplan. Then, DUNDIRESULT will - let you find out how many results there are, and access each one - without having to the query again. - - * include/asterisk/lock.h: Remove a message that goes to LOG_ERROR - that's not really an error. - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add a - min-announce-frequency option to queues.conf which allows you to - control the minimum amount of time between queue announcements - for use when the caller's queue position changes frequently. - (issue #9604, patch by Matthew Roth) - - * /, channels/chan_agent.c: Merged revisions 62218 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 - Apr 2007) | 11 lines Fix a weird problem where when a caller - talking to someone sitting behind an agent channel sent a digit, - the digit would be played to the agent for forever. This is - because chan_agent always returned -1 from its send_digit_begin - and _end callbacks. This non-zero return value indicates to the - Asterisk core that it would like an inband DTMF generator put on - the channel. However, this is the wrong thing to do. It should - *always* return 0, instead. When the digit begin and end - functions are called on the proxied channel, the underlying - channel will indicate whether inband DTMF is needed or not, and - the generator will be put on that one, and not the Agent channel. - (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed - by me) ........ - -2007-04-27 16:18 +0000 [r62175] Jason Parker <jparker@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 62174 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, - 27 Apr 2007) | 11 lines Merged revisions 62173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 - lines This transcoder message needn't be a NOTICE. I've seen it - cause confusion more than a few times. ........ ................ - -2007-04-27 16:15 +0000 [r62172] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: Merged revisions 62171 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | - 6 lines If no variables were passed into - pbx_substitute_variables_helper_full(), then don't even bother - creating a temporary bogus channel, since that is only for - allowing certain functions to operate on the variables as if they - were on a channel. Most importantly, this fixes a crash. (issue - #9613, reported by callguy, fixed by me) ........ - -2007-04-27 14:40 +0000 [r62096-62141] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #9545 Autocomplete for "sip - unregister" cli command. (eliel) Thanks! - - * /, channels/chan_sip.c: Merged revisions 62137 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, - 27 Apr 2007) | 12 lines Merged revisions 62126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 - lines Issue #7351 - SIP Cancel fails due to the wrong contact - uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - - THANKS!!!! THis was a hard one to catch. ........ - ................ - - * /: Blocking patch to 1.4 that was alredy in trunk - -2007-04-26 16:35 +0000 [r62039] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 62038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, - 26 Apr 2007) | 10 lines Merged revisions 62037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 - lines Revert previous fix for when the IAX2 channel goes funky - (that's the technical term). This is causing legit calls to be - prematurely hung up. (issue #9600 reported by justdave) ........ - ................ - -2007-04-26 03:24 +0000 [r62006] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 62005 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 - lines Missed an ast_app_group_discard during merge. Thanks - blitzrage! ........ - -2007-04-26 01:50 +0000 [r61960-61962] Joshua Colp <jcolp@digium.com> - - * /, res/res_monitor.c: Merged revisions 61961 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2 - lines Don't always say that the channel is being paused if it is - actually being unpaused in the Manager ack message. (reported by - jsmith in #asterisk-bugs) ........ - - * /, main/config.c: Merged revisions 61959 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61959 | file | 2007-04-25 21:27:18 -0400 (Wed, - 25 Apr 2007) | 10 lines Merged revisions 61958 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 - lines Don't count failed include attempts against the - configuration include level. (issue #9593 reported by mostyn) - ........ ................ - -2007-04-25 22:34 +0000 [r61915] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61914 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61914 | kpfleming | 2007-04-25 17:29:53 -0500 - (Wed, 25 Apr 2007) | 10 lines Merged revisions 61913 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 - Apr 2007) | 2 lines handle a very bizarre race condition with - channels being redirected before a simple switch can be started - on them (issue #9286) ........ ................ - -2007-04-25 22:01 +0000 [r61864-61876] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 61870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61870 | russell | 2007-04-25 16:59:07 -0500 - (Wed, 25 Apr 2007) | 10 lines Merged revisions 61866 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 - Apr 2007) | 2 lines If the callerid= option is specified, but - empty, clear any previous data. ........ ................ - - * /, channels/chan_iax2.c: Merged revisions 61863 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61863 | russell | 2007-04-25 16:13:15 -0500 - (Wed, 25 Apr 2007) | 10 lines Merged revisions 61862 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 - Apr 2007) | 2 lines Ensure that callerid settings are reset on a - reload. ........ ................ - -2007-04-25 19:27 +0000 [r61806] Joshua Colp <jcolp@digium.com> - - * main/channel.c, include/asterisk/app.h, funcs/func_groupcount.c, - /, main/app.c, main/cli.c: Merged revisions 61805 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, - 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 - lines Merge rewritten group counting support. No more storing - data on the variable list of the channels. That was bad, mmmk? - (issue #7497 reported by sabbathbh) ........ ................ - -2007-04-25 16:23 +0000 [r61788-61800] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61799 | russell | 2007-04-25 11:22:07 -0500 - (Wed, 25 Apr 2007) | 11 lines Merged revisions 61798 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 - Apr 2007) | 3 lines Fix a typo where cid_num got copied instead - of cid_ani. (issue #9587, reported and patched by xrg) ........ - ................ - - * main/manager.c, /: Merged revisions 61787 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61787 | russell | 2007-04-24 16:34:53 -0500 - (Tue, 24 Apr 2007) | 12 lines Merged revisions 61786 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 - Apr 2007) | 4 lines Don't crash if a manager connection provides - a username that exists in manager.conf but does not have a - password, and also requests MD5 authentication. (ASA-2007-012) - ........ ................ - -2007-04-24 19:08 +0000 [r61784] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c, /: removed #if 0 block from chan_zap - restart_monitor() - -2007-04-24 19:03 +0000 [r61775-61782] Russell Bryant <russell@digium.com> - - * main/channel.c, /, include/asterisk/channel.h: Merged revisions - 61781 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | - 6 lines Improve DTMF handling in ast_read() even more in response - to a discussion on the asterisk-dev mailing list. I changed the - enforced minimum length of a digit from 100ms to 80ms. - Furthermore, I made it now enforce a gap of 45ms in between - digits. These values are not configurable in a configuration file - right now, but they can be easily changed near the top of - main/channel.c. ........ - - * main/dial.c, /: Merged revisions 61774 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | - 5 lines Add a few more state changes in handle_frame_ownerless() - so that the SLA code will get notified of these changes even when - an owner channel is not provided. This isn't from a specific bug - report, it's just something I noticed while poking around. - ........ - -2007-04-24 16:10 +0000 [r61773] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61772 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, - 24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 - lines Allow RFC2833 to be sent in the response SDP when an INVITE - comes in without SDP. (issue #9546 reported by mcrawford) - ........ ................ - -2007-04-23 18:49 +0000 [r61760-61767] Russell Bryant <russell@digium.com> - - * main/manager.c: When building a JSON encoded string in the - GetConfigJSON manager action, escape the '\' and '"' characters. - (issue #9475, reported by pari, patch by me) - - * main/pbx.c, /: Merged revisions 61765 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | - 5 lines Some dialplan functions, such as CUT(), expect to operate - on variables on a channel. So, this little hack lets them work in - places where a channel doesn't exist, such as within DUNDi - configuration. (issue #9465, reported and patched by Corydon76, - testing by blitzrage) ........ - - * main/channel.c, /: Merged revisions 61763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | - 4 lines Ensure that digits passing through Asterisk have a - reasonable minimum length. It is currently 100 ms. If someone - thinks this should be different, feel free to speak up. (related - to issues #8944, #9250, and #9348) ........ - - * CHANGES: Add OSP support for IAX2 to the changes file. Also, - slightly reorganize some of the content. - -2007-04-20 21:37 +0000 [r61706-61708] Jason Parker <jparker@digium.com> - - * /, main/rtp.c: Merged revisions 61707 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 - lines Avoid invalid seqno cycling detection. Per comment from - Dave Troy: This adds back in some simple typecasting I had in an - earlier version which I realize now may be breaking things. Issue - #9554. ........ - - * /, main/loader.c: Merged revisions 61705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61705 | qwell | 2007-04-20 16:15:29 -0500 (Fri, - 20 Apr 2007) | 12 lines Merged revisions 61704 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 - lines Fix an issue that I noticed while looking over issue 9571. - The reload timestamp was getting set after reloading the built-in - stuff, and before the modules. ........ ................ - -2007-04-20 21:12 +0000 [r61698-61702] Russell Bryant <russell@digium.com> - - * channels/iax2-parser.h, funcs/func_channel.c, channels/iax2.h, - channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from - team/russell/iax2_osp This set of changes adds OSP support to - chan_iax2. However, I have modified the patch a bit from what was - submitted. You now use the CHANNEL() function to get and set the - OSP token for IAX2. (issue #8531, reported by and original patch - by homesick, patch updated by me) - - * /, main/rtp.c: Merged revisions 61697 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | - 2 lines Remove a stray debug message introduced by a recent - commit. ........ - -2007-04-20 19:54 +0000 [r61695] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 61694 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri, - 20 Apr 2007) | 13 lines Merged revisions 61692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 - lines If the '* to hangup' option is not enabled, we don't need - to disable * as a valid exit key. If it was enabled, this - statement would've never been checked in the first place. Issue - #9552 ........ ................ - -2007-04-20 18:23 +0000 [r61691] Russell Bryant <russell@digium.com> - - * main/manager.c, /, include/asterisk/config.h, main/config.c, - apps/app_voicemail.c: Merged revisions 61690 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) | - 4 lines Fix the UpdateConfig manager action to properly treat - "variables" and "objects" differently (a=b versus a=>b). (issue - #9568, reported by pari, patch by me) ........ - -2007-04-20 08:41 +0000 [r61689] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Use the last line in the SDP, even if it - has no CRLF. Remember Jon Postel :-) This code exists in 1.2 and - 1.4 but was removed from trunk for some unknown reason. - -2007-04-19 04:37 +0000 [r61682-61684] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /: Merged revisions 61683 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61683 | tilghman | 2007-04-18 23:36:20 -0500 (Wed, 18 Apr 2007) - | 2 lines Bug 9557 - simple reason why reading a function always - returned NULL ........ - - * funcs/func_groupcount.c, /, funcs/func_timeout.c, - funcs/func_cdr.c, funcs/func_callerid.c: Merged revisions 61681 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61681 | tilghman | 2007-04-18 21:45:05 -0500 - (Wed, 18 Apr 2007) | 13 lines Merged revisions 61680 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 - Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action - without a Channel parameter can cause Asterisk to crash. The - reason this needs to be fixed in the functions instead of in AMI - is because Channel can legitimately be NULL, such as when - retrieving global variables. ........ ................ - -2007-04-18 22:11 +0000 [r61679] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/Makefile: Merged revisions 61678 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61678 | kpfleming | 2007-04-18 17:10:23 -0500 (Wed, 18 Apr 2007) - | 2 lines allow external build systems to extract the required - sound file versions ........ - -2007-04-18 20:48 +0000 [r61671-61677] Olle Johansson <oej@edvina.net> - - * /, main/rtp.c: Merged revisions 61676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 - lines Clean upp formatting, add some doxygen stuff while we're in - cleaning mode... Thanks Kevin! ........ - - * /, main/rtp.c: Merged revisions 61674 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 - lines Issue #9554 - Improve RTCP (Dave Troy) ........ - - * apps/app_minivm.c (added), configs/extensions_minivm.conf.sample - (added), configs/minivm.conf.sample (added): Mini-voicemail - an - embryo for a new voicemail system based on building blocks - instead of one large monolithic app. Supports multiple templates - and is designed mostly for voicemail delivery over e-mail. - There's a todo with a list of ideas in the source code if you - want to contribute. Feedback is appreciated! - -2007-04-16 15:40 +0000 [r61667] Olle Johansson <oej@edvina.net> - - * include/asterisk/rtp.h: Doxygen changes - -2007-04-14 18:22 +0000 [r61661] Claude Patry <cpatry@gmail.com> - - * main/say.c: test my new trunk access ;) - -2007-04-13 21:23 +0000 [r61660] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_sip.c: added CLI 'sip unregister <peer>' for issue - 9326. thanks eliel - -2007-04-13 21:22 +0000 [r61659] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 61658 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61658 | murf | 2007-04-13 15:17:20 -0600 (Fri, 13 Apr 2007) | 1 - line This is a fix to the way CDR merge handles the data that - results from ForkCDR. ........ - -2007-04-13 19:18 +0000 [r61649-61657] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 61656 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, - 13 Apr 2007) | 10 lines Merged revisions 61655 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 - lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves - the same as OUTBOUND_GROUP except it will get unset after use so - it won't get accidentally inherited. (issue #BE-140) ........ - ................ - - * /, apps/app_speech_utils.c: Merged revisions 61651 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr - 2007) | 2 lines Do not bother looking for a result if none are - present. ........ - - * /, channels/chan_sip.c: Merged revisions 61648 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 - lines For those very verbose SIP implementations that attach tons - of info to the Contact header... let's increase our variable - sizes. (issue #9535 reported by jeffg) ........ - -2007-04-13 17:15 +0000 [r61647] Russell Bryant <russell@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 61645 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61645 | russell | 2007-04-13 12:10:19 -0500 (Fri, 13 Apr 2007) | - 3 lines Eliminate a compiler warning with ODBC_STORAGE enabled so - that it will build under dev-mode. ........ - -2007-04-13 17:11 +0000 [r61646] Steve Murphy <murf@digium.com> - - * /, channels/chan_oss.c: Merged revisions 61644 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 - line A fix for chan_oss that resulted from the CDR changes; it - helps to use the right info. ........ - -2007-04-13 16:35 +0000 [r61618-61642] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61641 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 - lines Don't assume the callid of a dialog will be set, as in some - circumstances it may not. (issue #9534 reported by tecnoxarxa) - ........ - - * channels/chan_sip.c: Don't treat a host lookup as failed if - sipregs is not in use when doing a realtime lookup. (issue #9255 - reported by sergee) - -2007-04-11 22:19 +0000 [r61575-61599] Dwayne M. Hubbard <dhubbard@digium.com> - - * doc/asterisk-conf.tex: clarified 'minmemfree' description in - doc/asterisk-conf.tex - - * main/asterisk.c, doc/asterisk-conf.tex: fixed the '-e' command - line option for minmemfree. updated doc/asterisk-conf.tex - - * main/pbx.c, include/asterisk/options.h, main/asterisk.c: changed - #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO) - - * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added - HAVE_SYSINFO preprocessor directives for portability and general - happiness - -2007-04-11 20:21 +0000 [r61557] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: Add a - configure script check for sysinfo support. - -2007-04-11 19:11 +0000 [r61539] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added - option_minmemfree for use in asterisk.conf to specify the amount - of minimum free memory prior to accepting calls. added CLI 'core - show sysinfo' to display system information - -2007-04-11 17:07 +0000 [r61522] Joshua Colp <jcolp@digium.com> - - * main/logger.c: Output verbose messages to the normal logger as - well. (issue #9476 reported by gdalgliesh) - -2007-04-11 16:06 +0000 [r61478] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61477 | russell | 2007-04-11 11:05:29 -0500 - (Wed, 11 Apr 2007) | 13 lines Merged revisions 61476 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 - Apr 2007) | 5 lines If someone sets the "useragent" option in - sip.conf to be empty, then don't add the User-Agent header at - all. It is an optional header, anyway. Also, the bug report says - that some of Japan's SIP providers don't allow it for some weird - reason. (issue #9488, reported by makoto, fixed by me) ........ - ................ - -2007-04-11 15:48 +0000 [r61460] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_lib.c: Merged revisions - 61342,61372-61373,61443 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 - lines AOCD's are now exported to asterisk channel variables. - ........ r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr - 2007) | 2 lines Ignore facility messages in case we don't have a - corresponding channel object. ........ r61373 | nadi | 2007-04-11 - 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines Export AOCD variables - on misdn_hangup. ........ r61443 | nadi | 2007-04-11 17:39:14 - +0200 (Mi, 11 Apr 2007) | 2 lines Don't export AOCD variables on - misdn_hangup anymore, this was mainly a fix for trunk.. ........ - -2007-04-11 15:25 +0000 [r61379-61429] Russell Bryant <russell@digium.com> - - * funcs/func_devstate.c: Add a minor loop optimization to the - custom device state callback. Once the correct device is found, - it should just break out of the loop ... - - * /, channels/chan_sip.c: Merged revisions 61427 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61427 | russell | 2007-04-11 10:09:39 -0500 - (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 - Apr 2007) | 6 lines Fix a bug with switching between host=dynamic - and using specific hosts for peers. The code would only reset the - peer's address when it is dynamic if it was a new peer structure. - Now, it will also reset the address if it was already in the peer - list, but before the reload, it was not dynamic. (issue #9515, - reported by caio1982, fixed by me) ........ ................ - - * /, main/http.c: Merged revisions 61407 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61407 | russell | 2007-04-11 09:48:01 -0500 (Wed, 11 Apr 2007) | - 4 lines Add "svgz" to the mimetypes table. (issue #9510, bkruse) - In passing, constify the elements of the mimetypes table. - ........ - - * /, channels/chan_sip.c: Merged revisions 61377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61377 | russell | 2007-04-11 09:04:44 -0500 - (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 - Apr 2007) | 5 lines Remove the attempt at reporting configuration - errors in sip.conf. This can cause a bunch of improper messages - when using realtime. I give up. As oej tried to convince me when - I put this in, there is just no easy way to do it. (inspired by a - message on the -dev list) ........ ................ - -2007-04-11 14:09 +0000 [r61378] Steve Murphy <murf@digium.com> - - * apps/app_voicemail.c: via 8119, a patch to allow voicemail data - to be stored in RealTime. - -2007-04-11 14:01 +0000 [r61375] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Remove duplicate prototype declaration. - (issue #9517 reported by junky) - -2007-04-11 13:41 +0000 [r61374] Steve Murphy <murf@digium.com> - - * include/asterisk/config.h, main/config.c: via 8118, a RealTime - upgrade to make RT a complete storage abstraction. The - store/destroy mechanisms needed these missing peices. - -2007-04-10 23:55 +0000 [r61324] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, main/manager.c, configs/manager.conf.sample, - include/asterisk/manager.h: Issue 6082 - New DTMF event for - manager - -2007-04-10 22:02 +0000 [r61303] Doug Bailey <dbailey@digium.com> - - * channels/chan_zap.c: Added zapata.conf parameter "cid_rxgain" to - allow the user to adjust the gain bump used during CID - acquisition. - -2007-04-10 20:50 +0000 [r61222-61283] Russell Bryant <russell@digium.com> - - * CHANGES: Note the bridge manager action and application in the - CHANGES file. - - * res/res_features.c: Merge changes from team/russell/issue_5841: - This patch adds a "Bridge" Manager action, as well as a "Bridge" - dialplan application. The manager action will allow you to steal - two active channels in the system and bridge them together. Then, - the one that did not hang up will continue in the dialplan. Using - the application will bridge the calling channel to an arbitrary - channel in the system. Whichever channel does not hang up here - will continue in the dialplan, as well. This patch has been - touched by a bunch of people over the course of a couple years. - Please forgive me if I have missed your name in the history of - things. The most recent patch came from issue #5841, but there is - also a reference to an earlier version of this patch from issue - #4297. The people involved in writing and/or reviewing the code - include at least: twisted, mflorrel, heath1444, davetroy, - tim_ringenbach, moy, tmancill, serge-v, and me. There are also - positive test reports from many people. - - * main/dial.c, include/asterisk/dial.h: Add an option to the dial - API for playing music instead of ringing to the caller. I started - this for use with SLA but ended up deciding not to use it. - However, there is no reason not to put this part in, anyway. - -2007-04-10 16:07 +0000 [r61221] Steve Murphy <murf@digium.com> - - * channels/chan_jingle.c: updated ast_channel_alloc() call to - include the 4 extra args everyone got. Not much info there, as - the config file evidently does not allow amaflags, or accountcode - settings; and the pvt's exten doesn't sound like what we need in - the cdr, either. - -2007-04-10 12:47 +0000 [r61184] Nadi Sarrar <ns@beronet.com> - - * /, channels/misdn_config.c: Merged revisions 61183 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, - 10 Apr 2007) | 10 lines Merged revisions 61170 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 - lines msns config parameter defaults to '*' ........ - ................ - -2007-04-10 05:41 +0000 [r61152] Steve Murphy <murf@digium.com> - - * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc, - channels/chan_zap.c, /, channels/chan_sip.c, res/res_features.c, - channels/chan_agent.c, include/asterisk/channel.h, - channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, - main/channel.c, main/cdr.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, - apps/app_cdr.c, apps/app_voicemail.c: Merged revisions 60989 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 - line This is a big improvement over the current CDR fixes. It may - still need refinement, but this won't have as many folks - bothered. This also adds the mods from 1.4/r.61136; ........ - -2007-04-09 22:49 +0000 [r61116] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: Remove unused instances of unnamed enums. - -2007-04-09 20:01 +0000 [r61073] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 61072 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, - 09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 - lines - Don't send ActionID before Response: header. - Don't use - a blank in an AMI header ........ ................ - -2007-04-09 19:57 +0000 [r61065-61071] Kevin P. Fleming <kpfleming@digium.com> - - * main/minimime/mm_envelope.c, /: Merged revisions 61070 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61070 | kpfleming | 2007-04-09 14:55:14 -0500 (Mon, 09 Apr 2007) - | 2 lines fix up some warnings found using --enable-dev-mode - ........ - - * /, main/minimime/tests/CVS (removed), main/minimime/Doxyfile - (removed), main/minimime/tests/messages/CVS (removed): Merged - revisions 61062 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61062 | kpfleming | 2007-04-09 14:49:09 -0500 (Mon, 09 Apr 2007) - | 2 lines remove some more stuff we don't need ........ - -2007-04-09 19:06 +0000 [r61023] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 61022 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r61022 | qwell | 2007-04-09 14:05:48 -0500 (Mon, 09 Apr 2007) | 4 - lines Use the appropriate interface name with COMPLETECALLER. - Issue 9395. ........ - -2007-04-09 19:05 +0000 [r60985-61021] Olle Johansson <oej@edvina.net> - - * main/manager.c: Add hint to ExtensionStatus AMI event in manager - - * channels/chan_sip.c, CHANGES, channels/chan_iax2.c: use - "ChannelType" in events to indicate which channel driver that - generates the event. This replaces "ChannelDriver" and "Channel", - previously used to indicate channel driver. ChannelType is more - in line with "core show channeltypes" - - * res/res_jabber.c: Fix JabberEvents - - * /, res/res_jabber.c: Fix missing newline in JabberEvent - -2007-04-09 17:23 +0000 [r60937] Jason Parker <jparker@digium.com> - - * /, apps/app_directory.c: Merged revisions 60936 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60936 | qwell | 2007-04-09 12:22:59 -0500 (Mon, - 09 Apr 2007) | 13 lines Merged revisions 60935 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 - lines Allow matching on names shorter than 3 chars. This also - fixes the case where somebody wants to match on less then 3 - chars. Issue 9071 ........ ................ - -2007-04-09 16:30 +0000 [r60917] Dwayne M. Hubbard <dhubbard@digium.com> - - * UPGRADE.txt: updated UPGRADE.txt to include format_wav changes - -2007-04-09 12:33 +0000 [r60898] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make RTP session ID and session version - generation random. (issue #9456 reported by tjardick) - -2007-04-09 03:04 +0000 [r60848-60851] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk.h, /, main/asterisk.c: Merged revisions 60850 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500 - (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 - Apr 2007) | 2 lines Don't check for error when lowering priority - (according to the manpage, it should never happen anyway). It - might could happen, though, if another thread messed with the - priority, so safeguard against that (reported via -dev list). - ........ ................ - - * channels/chan_local.c, /: Merged revisions 60847 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60847 | tilghman | 2007-04-08 21:42:48 -0500 - (Sun, 08 Apr 2007) | 10 lines Merged revisions 60846 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 - Apr 2007) | 2 lines Bug 9505 - If the return value for - local_queue_frame is set, then p->lock is no longer valid. - ........ ................ - -2007-04-09 01:06 +0000 [r60763-60799] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 60798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60798 | file | 2007-04-08 21:03:14 -0400 (Sun, - 08 Apr 2007) | 10 lines Merged revisions 60797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 - lines When calling a device that then forwards us elsewhere... we - have to make our channels compatible if it is the only channel - being dialed. (issue #9445 reported by marcelbarbulescu) ........ - ................ - - * channels/chan_sip.c: Add counter for sip show registry CLI - command. (issue #9352 reported by junky) - - * /, apps/app_queue.c: Merged revisions 60762 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60762 | file | 2007-04-08 13:04:44 -0400 (Sun, 08 Apr 2007) | 2 - lines Allow app_queue to use MONITOR_EXEC even if MONITOR_OPTIONS - is not set. (issue #9495 reported by cduffy) ........ - -2007-04-08 14:23 +0000 [r60662-60715] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_macro.c: Merged revisions 60713 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60713 | tilghman | 2007-04-08 09:14:29 -0500 - (Sun, 08 Apr 2007) | 10 lines Merged revisions 60711 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 - Apr 2007) | 2 lines Gosub called within a Macro resets the - arguments improperly and causes general weirdness. (Issue 8329) - ........ ................ - - * /, formats/format_wav.c, main/http.c: Merged revisions 60712 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60712 | tilghman | 2007-04-08 09:12:00 -0500 (Sun, 08 Apr 2007) - | 2 lines Fix --enable-dev-mode ........ - - * /, main/file.c: Merged revisions 60661 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60661 | tilghman | 2007-04-07 20:40:47 -0500 - (Sat, 07 Apr 2007) | 10 lines Merged revisions 60660 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 - Apr 2007) | 2 lines Bug 9486 - memory leak when opening a - filestream ........ ................ - -2007-04-06 22:29 +0000 [r60641] Dwayne M. Hubbard <dhubbard@digium.com> - - * formats/format_wav.c: removed GAIN preprocessor definition, - removed needsgain from struct wav_desc, removed unnecessary gain - code from wav_read() and wav_write() - -2007-04-06 21:43 +0000 [r60566-60623] Russell Bryant <russell@digium.com> - - * main/minimime/Makefile: Filter out -Wundef so that the - automatically generated C files will compile cleanly - - * main/minimime/mytest_files (removed), main/minimime/sys/CVS - (removed), main/minimime/.cvsignore (removed), - main/minimime/mm-docs (removed), main/minimime/test (removed): - Remove a bunch of files that weren't supposed to get added. - - * main/minimime/mm-docs/html/mm__envelope_8c.html, - main/minimime/tests/messages, include/asterisk/autoconfig.h.in, - main/minimime/mm-docs/html/mm__context_8c.html, - main/minimime/sys, main/minimime/tests/Makefile, - main/minimime/tests/CVS/Root, main/minimime/sys/CVS/Entries, - main/minimime/mm-docs/latex/mm__mimeutil_8c.tex, configure, - main/strcompat.c, main/http.c, main/minimime/mm_error.c, - main/minimime/mm-docs/html/globals_func.html, - main/minimime/mm-docs/html/group__mimeutil.html, - main/minimime/mm-docs/latex/doxygen.sty, - main/minimime/mm_param.c, main/minimime/test/CVS, configure.ac, - main/minimime/.cvsignore, main/minimime/mm_init.c, - main/minimime/mm-docs/html/mm__queue_8h-source.html, - main/minimime/mm-docs/html/mm__error_8c.html, - main/minimime/mm-docs/html/tabs.css, main/minimime/mm_envelope.c, - main/minimime/mimeparser.h, main/minimime/mimeparser.l, - main/minimime/mm_context.c, - main/minimime/mm-docs/html/group__mimepart.html, - main/minimime/mm-docs/latex/group__envelope.tex, - main/minimime/tests/messages/CVS, - main/minimime/mm-docs/html/mm__contenttype_8c.html, - main/minimime/mm-docs/html/pages.html, - main/minimime/mm-docs/html/group__error.html, - main/minimime/mm-docs/latex/group__context.tex, - main/minimime/mimeparser.y, Makefile.moddir_rules, - main/minimime/sys/mm_queue.h, - main/minimime/mm-docs/html/bug.html, - main/minimime/mm-docs/html/mimeparser_8tab_8h-source.html, - main/minimime/tests/messages/CVS/Root, - main/minimime/mm_mimepart.c, - main/minimime/mm-docs/latex/Makefile, - main/minimime/mm_internal.h, main/minimime/tests/CVS, - main/minimime/mm-docs/latex/mm__param_8c.tex, - main/minimime/tests/parse.c, main/minimime/mm_base64.c, - main/minimime/mm.h, main/minimime/mm_header.c, - main/minimime/mm-docs/latex/mm__parse_8c.tex, - main/minimime/mm-docs/html/mimeparser_8h-source.html, - main/minimime/mm-docs/html/files.html, - main/minimime/mm-docs/latex/mm__contenttype_8c.tex, - main/minimime/mm-docs/html/mm__mem_8h-source.html, - main/minimime/mm_codecs.c, - main/minimime/mm-docs/latex/mm__mimepart_8c.tex, - main/minimime/mytest_files/mytest.c, - main/minimime/mm-docs/html/mm__mimeutil_8c.html, - main/minimime/mm-docs/latex/files.tex, - main/minimime/test/CVS/Entries, - main/minimime/mm-docs/latex/modules.tex, - main/minimime/tests/messages/CVS/Repository, - configs/http.conf.sample, main/minimime/mm_contenttype.c, - main/minimime/tests/messages/test1.txt, - main/minimime/mm-docs/html/mm__param_8c.html, - main/minimime/tests/messages/test3.txt, - main/minimime/tests/messages/test5.txt, - main/minimime/tests/messages/test7.txt, - main/minimime/mm-docs/html/group__contenttype.html, - main/minimime/mm-docs, - main/minimime/mytest_files/ast_postdata3.gz, main/minimime - (added), main/minimime/Make.conf, - main/minimime/mm-docs/latex/group__contenttype.tex, - main/minimime/mm_warnings.c, main/minimime/mm_queue.h, - main/minimime/mm-docs/html/mm__util_8c.html, - main/minimime/mm-docs/html/doxygen.css, /, - main/minimime/mm-docs/html/mm__internal_8h.html, - main/minimime/tests/messages/CVS/Entries, main/minimime/Doxyfile, - main/minimime/minimime.c, main/minimime/mimeparser.yy.c, - main/minimime/tests/CVS/Entries.Log, main/minimime/test.sh, - include/asterisk/compat.h, main/minimime/test/CVS/Repository, - main/minimime/mm_mimeutil.c, main/minimime/tests, - main/minimime/mm-docs/latex/group__mimepart.tex, - main/minimime/tests/CVS/Entries, main/Makefile, - main/minimime/mm-docs/latex/mm__envelope_8c.tex, - main/minimime/mm-docs/latex/mm__util_8c.tex, - main/minimime/mm-docs/latex/pages.tex, - main/minimime/mm-docs/latex/group__mimeutil.tex, - main/minimime/mm-docs/latex, - main/minimime/mm-docs/html/mm_8h-source.html, - main/minimime/Makefile, - main/minimime/mm-docs/latex/mm__internal_8h.tex, - main/minimime/mm-docs/refman.pdf, include/asterisk/manager.h, - main/minimime/mm-docs/latex/mm__context_8c.tex, - main/minimime/mm-docs/latex/group__param.tex, - main/minimime/mm-docs/latex/group__codecs.tex, - main/minimime/tests/create.c, main/minimime/mm_util.c, - main/minimime/mm-docs/latex/bug.tex, - main/minimime/mimeparser.tab.c, main/minimime/mm_util.h, - main/minimime/mytest_files/ast_postdata, - main/minimime/mm-docs/html/group__envelope.html, - main/minimime/mm-docs/html/group__util.html, - main/minimime/mimeparser.tab.h, - main/minimime/mm-docs/html/mm__parse_8c.html, - main/minimime/mm-docs/html, - main/minimime/mm-docs/latex/group__util.tex, - main/minimime/mm-docs/html/group__context.html, - main/minimime/mm-docs/html/mm__internal_8h-source.html, - main/minimime/mytest_files, - main/minimime/mm-docs/html/mm__util_8h-source.html, - main/minimime/sys/CVS, - main/minimime/mm-docs/html/group__codecs.html, main/manager.c, - main/minimime/sys/CVS/Repository, - main/minimime/mm-docs/html/globals.html, - main/minimime/mm-docs/html/mm__mimepart_8c.html, - main/minimime/tests/CVS/Repository, - main/minimime/mm-docs/html/index.html, - main/minimime/mm-docs/html/modules.html, main/minimime/test, - main/minimime/mytest_files/ast_postdata2, - main/minimime/mm-docs/latex/group__error.tex, - main/minimime/mm-docs/html/mm__header_8c.html, - main/minimime/strlcpy.c, - main/minimime/mm-docs/html/group__param.html, - main/minimime/mm-docs/latex/refman.tex, main/minimime/mm_parse.c, - main/minimime/mm-docs/latex/mm__header_8c.tex, - main/minimime/mm-docs/latex/mm__error_8c.tex, - main/minimime/mm_mem.c, - main/minimime/mm-docs/html/mm__codecs_8c.html, - main/minimime/tests/messages/test2.txt, - main/minimime/tests/messages/test4.txt, - main/minimime/sys/CVS/Root, - main/minimime/tests/messages/test6.txt, - main/minimime/test/CVS/Root, main/minimime/strlcat.c, - main/minimime/mm_mem.h, - main/minimime/mm-docs/latex/mm__codecs_8c.tex: Merged revisions - 60603 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | - 13 lines To be able to achieve the things that we would like to - achieve with the Asterisk GUI project, we need a fully functional - HTTP interface with access to the Asterisk manager interface. One - of the things that was intended to be a part of this system, but - was never actually implemented, was the ability for the GUI to be - able to upload files to Asterisk. So, this commit adds this in - the most minimally invasive way that we could come up with. A lot - of work on minimime was done by Steve Murphy. He fixed a lot of - bugs in the parser, and updated it to be thread-safe. The ability - to check permissions of active manager sessions was added by - Dwayne Hubbard. Then, hacking this all together and do doing the - modifications necessary to the HTTP interface was done by me. - ........ - - * /, apps/app_meetme.c: Merged revisions 60565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60565 | russell | 2007-04-06 14:50:52 -0500 (Fri, 06 Apr 2007) | - 3 lines When a station picks up a trunk that was on hold, make - the hints reflect that nobody has the trunk on hold anymore. - ........ - -2007-04-06 19:26 +0000 [r60531] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Use the same parameter to the two "Registry" - AMI events - ChannelDriver - -2007-04-06 18:59 +0000 [r60522] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 60521 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) | - 16 lines Fix a few problems with SLA. (issue #9459, reported by - francesco_r, fixed by me) * The original behavior was that if one - station put a call on hold, another one picked it up, and then - hung up, the code would still consider the call on hold by the - first station, so the trunk would not be hung up. However, to - better comply with what most people seem to expect it to behave, - it will now hang up the trunk. * Fix a problem with "barge=no". - This was only intended to prevent people from joining calls that - are in progress. However, it also prevented other people from - picking up a call that was on hold. This has been fixed. * When - there are no active stations on a trunk and it is on hold, the - code now indicates the HOLD and UNHOLD conditions to the trunk - channel. This allows music on hold to be played to the trunk when - it is on hold. ........ - -2007-04-06 18:26 +0000 [r60486-60487] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 60485 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2 - lines Make sure we check the faxdetect option before doing fax - processing ........ - - * channels/chan_zap.c, /: Merged revisions 60459 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri, - 06 Apr 2007) | 10 lines Merged revisions 60456 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 - lines There should only be one code path for doing DTMF - conditionals on channels. This fixes it. ........ - ................ - -2007-04-06 14:53 +0000 [r60400] Kevin P. Fleming <kpfleming@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 60399 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60399 | kpfleming | 2007-04-06 09:49:51 -0500 - (Fri, 06 Apr 2007) | 10 lines Merged revisions 60398 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 - Apr 2007) | 2 lines remove undocumented 'cardsmode' parameter and - stop searching for transcoders during reload() ........ - ................ - -2007-04-06 01:29 +0000 [r60362-60363] Joshua Colp <jcolp@digium.com> - - * include/asterisk/speech.h, res/res_speech.c: Major res_speech - cleanup. It looks much better now! - - * /, include/asterisk/speech.h, res/res_speech.c, - apps/app_speech_utils.c: Merged revisions 60361 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 - lines Add support for returning different types of results (ie: - NBest). ........ - -2007-04-05 23:08 +0000 [r60326] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, formats/format_wav.c: Merged revisions 60325 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60325 | dhubbard | 2007-04-05 17:58:01 -0500 (Thu, 05 Apr 2007) - | 1 line modified default GAIN for issue 5823, thanks jrwalliker - ........ - -2007-04-05 22:40 +0000 [r60324] Steve Murphy <murf@digium.com> - - * configs/cdr_custom.conf.sample, /, configs/cdr.conf.sample: - Merged revisions 60323 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1 - line Added some clarification to the example configs for CDRs, on - how to select a backend. Also, made cdr-csv the default if you - 'make samples', and no other changes. ........ - -2007-04-05 16:11 +0000 [r60269] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 60268 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60268 | qwell | 2007-04-05 11:10:48 -0500 (Thu, - 05 Apr 2007) | 13 lines Merged revisions 60267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 - lines Just because we can't find the voicemail configuration - file, doesn't mean that the module failed to load. The user could - be using realtime. Issue #9473 ........ ................ - -2007-04-05 15:48 +0000 [r60266] Russell Bryant <russell@digium.com> - - * /, main/http.c: Merged revisions 60265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60265 | russell | 2007-04-05 10:47:17 -0500 (Thu, 05 Apr 2007) | - 2 lines Add the MIME type for gif by request from Pari ........ - -2007-04-05 12:57 +0000 [r60215] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 60214 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, - 05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 - lines Only unlock our pvt and net locks if we are actually going - to try to lock the owner again. (issue #9472 reported by zoa) - ........ ................ - -2007-04-04 23:45 +0000 [r60193] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/callerid.c: ast_shrink_phone_number() must ignore - whitespace, otherwise my CIDCO callerid box gets LINE ERROR - -2007-04-04 17:41 +0000 [r60011-60141] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 60137 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60137 | russell | 2007-04-04 12:40:10 -0500 - (Wed, 04 Apr 2007) | 14 lines Merged revisions 60134 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 - Apr 2007) | 6 lines It is valid to redirect channels via the - manager interface that are not in the UP state. Instead of - checking for that to prevent to ensure a dead channel doesn't get - redirected, just use the ast_check_hangup() API call. (issue - #9457, reported by Callmewind, patch by me) (related to issue - #8977) ........ ................ - - * /, channels/chan_sip.c: Merged revisions 60112 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | - 3 lines Add a Content-Length of 0 to the response built by - transmit_response_with_unsupported(). (issue #9454, reported by - makoto, fixed by me) ........ - - * /, channels/chan_sip.c: Merged revisions 60088 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r60088 | russell | 2007-04-04 11:39:04 -0500 - (Wed, 04 Apr 2007) | 12 lines Merged revisions 60083 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 - Apr 2007) | 4 lines Fix the return value of - handle_common_options() so that it always properly indicates - whether it handled the option or not. (issue #9455, reported by - Netview, fixed by me) ........ ................ - - * /, apps/app_meetme.c: Merged revisions 60069 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) | - 4 lines Fix a problem where if a trunk was hung up while it was - on hold, all of the hints would reflect the line still on hold, - even though it should reflect that it is back to not in use. - (issue #9459, reported by francesco_r, fixed by me) ........ - - * channels/chan_jingle.c, channels/chan_gtalk.c, - doc/rtp-packetization.txt: Add support for RTP packetization in - chan_jingle and chan_gtalk. (issue #9416, phsultan) - -2007-04-03 19:43 +0000 [r59969] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 59963 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr - 2007) | 2 lines Don't clash when a person both speaks and uses - DTMF. ........ - -2007-04-03 19:17 +0000 [r59854-59940] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59939 | russell | 2007-04-03 14:16:53 -0500 - (Tue, 03 Apr 2007) | 12 lines Merged revisions 59938 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 - Apr 2007) | 4 lines Don't attempt to report configuration errors - in build_user(). oej pointed out that for a "friend" entry, this - won't work, because all user options are valid for peers, but not - the other way around. ........ ................ - - * /, channels/chan_sip.c: Merged revisions 59936 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59936 | russell | 2007-04-03 13:55:57 -0500 - (Tue, 03 Apr 2007) | 11 lines Merged revisions 59916 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 - Apr 2007) | 3 lines Make chan_sip report when it encounters an - unknown option. (issue #9440, reported by nightcrawler) ........ - ................ - - * channels/chan_sip.c: Remove a duplicate function prototype. - (issue #9444, junky) - - * /, main/app.c: Merged revisions 59887 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59887 | russell | 2007-04-03 13:01:49 -0500 - (Tue, 03 Apr 2007) | 13 lines Merged revisions 59886 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 - Apr 2007) | 5 lines When doing a built-in blind or attended - transfer, restore the ability to use '#' to terminate the number - and immediately do the transfer instead of having to dial the - number and just wait for the feature digit timeout. (issue #8366, - xueliangliang) ........ ................ - - * Makefile, /: Merged revisions 59853 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59853 | russell | 2007-04-03 11:03:35 -0500 (Tue, 03 Apr 2007) | - 1 line Ensure that menuselect gets executed in dependency check - mode every time you run make. ........ - -2007-04-03 11:15 +0000 [r59805] Nadi Sarrar <ns@beronet.com> - - * /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c: - Merged revisions 59804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, - 03 Apr 2007) | 15 lines Merged revisions 59788,59803 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr - 2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a - port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58 - +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the - 4th. ........ ................ - -2007-04-02 19:01 +0000 [r59725] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 59724 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59724 | file | 2007-04-02 14:58:24 -0400 (Mon, - 02 Apr 2007) | 10 lines Merged revisions 59723 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 - lines Increase the maximum size for a string of mailboxes to - 1024. (issue #9270 reported by rtucker) ........ ................ - -2007-04-02 17:40 +0000 [r59693] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: This hashing code is still causing some - random crashes on my system, and probably others, too. I don't - really have time to work on it at the moment, so I am just going - to revert it for now. - -2007-04-02 17:38 +0000 [r59692] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 59688 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59688 | murf | 2007-04-02 11:31:32 -0600 (Mon, 02 Apr 2007) | 1 - line continue in for-loop should go to the incrementer, not the - test. As per 9435, thanks to marcelbarbulescu ........ - -2007-04-02 16:08 +0000 [r59655] Russell Bryant <russell@digium.com> - - * /, main/netsock.c: Merged revisions 59654 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59654 | russell | 2007-04-02 10:39:07 -0500 - (Mon, 02 Apr 2007) | 14 lines Merged revisions 59608 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 - Apr 2007) | 6 lines Add the SO_REUSEADDR flag to sockets handled - by netsock. This is needed by the patch that went in for issue - 7874. chan_iax2 needs to be able to create socket that is - lisetning on INADDR_ANY, but also be able to bind sockets to - specific addresses. (Thanks to Stevenson on the asterisk-dev - mailing list for explaining why this flag was needed.) ........ - ................ - -2007-03-30 22:54 +0000 [r59574] Jason Parker <jparker@digium.com> - - * /, configure, main/Makefile, acinclude.m4: Merged revisions 59573 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2 - lines Add linux-uclibc host arch..."thingy". Sorry, I don't know - what it's called... ........ - -2007-03-30 20:54 +0000 [r59555] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Update to support multiple CIC groups and - DPCs per linkset. - -2007-03-30 17:57 +0000 [r59453-59523] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c, - include/asterisk/cdr.h: Merged revisions 59522 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 - line several changes via kpflemings review ........ - - * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c, - include/asterisk/cdr.h: Merged revisions 59486 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 - line These mods fix CDR issues from 8221, 8593, 8680, 8743, and - perhaps others. Mainly with CDRs generated from transfer - situations. ........ - - * /, configs/extensions.conf.sample: Merged revisions 59452 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1 - line A small clarification to keep bugs from being filed, and - confusion from rising, if clearglobalvars is set, and globals are - set in the AEL file. (9419) ........ - -2007-03-29 23:27 +0000 [r59364-59433] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Reduce the ridiculous number of variables - used in the load_config() function by just having one that can be - re-used. There is no functional change here (that is intentional, - anyway!). - - * CHANGES, apps/app_voicemail.c: Add the ability for the "voicemail - show users" CLI command to show users configured in realtime. - - * channels/chan_iax2.c: Fix an issue with hashing iax2 pvt - structures that caused random crashes on systems under heavy load - such as IAXtel. (possibly related to issue #9403) - - * /, res/res_jabber.c: Merged revisions 59363 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) | - 6 lines When building a response to a subscription, the "from" - must be the full Jabber ID. This fixes some problems where jabber - users are not able to add their Asterisk account to their user - list, since they are unable to get Asterisk to approve their - subscription. (issue #8210, reported by caspy, and verified by - bradtem) ........ - -2007-03-29 17:42 +0000 [r59362] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 59361 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59361 | file | 2007-03-29 13:38:55 -0400 (Thu, - 29 Mar 2007) | 10 lines Merged revisions 59360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 - lines Keep a global array of variables indicating whether certain - conference rooms are in use. This ensures that two people going - into a new dynamic conference when the 'e' option is set don't go - into the same conference room. (issue #8835 reported by eliel) - ........ ................ - -2007-03-29 17:20 +0000 [r59305-59359] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: Merged revisions 59358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59358 | russell | 2007-03-29 12:17:41 -0500 - (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 - Mar 2007) | 5 lines If an error occurs when reading from an RTP - socket, and the error code does not indicate that we should try - again, then return NULL instead of a "null frame". This will - prevent Asterisk from trying over and over again, and eventually - causing the system to crash. (issue #8285, john) ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 59341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | - 8 lines When the IAX2 read callback gets called, return NULL - instead of a "null frame". This will cause Asterisk to hangup the - call instead of keep trying whatever it was doing. Under normal - conditions, this function would *never* be called. However, the - author of this patch says an error will occur that will cause it - to get called every 100 thousand calls or so. When this does - happen, it puts the channel in a loop that eventually brings down - the system. So, hangup up the call is certainly a better - alternative. (issue #8286, john) ........ - - * Makefile, /: Merged revisions 59304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59304 | russell | 2007-03-29 11:25:41 -0500 (Thu, 29 Mar 2007) | - 2 lines Export the GTK2 library and include information to sub - Makefiles. ........ - -2007-03-29 16:08 +0000 [r59303] Tilghman Lesher <tlesher@digium.com> - - * /, cdr/cdr_odbc.c: Merged revisions 59302 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59302 | tilghman | 2007-03-29 11:07:05 -0500 - (Thu, 29 Mar 2007) | 11 lines Merged revisions 59301 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 - Mar 2007) | 3 lines Issue 9415 - No point to getting a diagnostic - field if we aren't doing anything with the information. (Plus, it - tends to crash the Postgres ODBC driver.) ........ - ................ - -2007-03-28 03:40 +0000 [r59290] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 59289 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59289 | tilghman | 2007-03-27 22:38:09 -0500 (Tue, 27 Mar 2007) - | 2 lines Another crash that I thought we had fixed already - - Issue 9396 ........ - -2007-03-28 00:09 +0000 [r59286] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c: added filtering options to 'zap show - channels' to implement functionality described in issue 6520 - -2007-03-27 23:38 +0000 [r59282-59285] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 59284 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59284 | tilghman | 2007-03-27 18:37:31 -0500 - (Tue, 27 Mar 2007) | 10 lines Merged revisions 59283 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 - Mar 2007) | 2 lines Oops ........ ................ - - * /, apps/app_voicemail.c: Merged revisions 59281 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59281 | tilghman | 2007-03-27 18:32:46 -0500 - (Tue, 27 Mar 2007) | 10 lines Merged revisions 59280 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 - Mar 2007) | 2 lines Fix a few remaining bad mmap(2) return values - ........ ................ - -2007-03-27 23:22 +0000 [r59274-59279] Russell Bryant <russell@digium.com> - - * /, apps/app_directory.c: Merged revisions 59278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59278 | russell | 2007-03-27 18:20:22 -0500 - (Tue, 27 Mar 2007) | 11 lines Merged revisions 59277 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 - Mar 2007) | 3 lines Fix the check of the return value from - mmap(). Thanks to Corydon for catching this one. ........ - ................ - - * /, apps/app_directory.c: Merged revisions 59275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59275 | russell | 2007-03-27 18:16:27 -0500 (Tue, 27 Mar 2007) | - 3 lines Fix app_directory to actually compile with ODBC_STORAGE, - and update the code to the latest res_odbc API. ........ - - * /, apps/Makefile: Merged revisions 59273 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59273 | russell | 2007-03-27 18:02:12 -0500 (Tue, 27 Mar 2007) | - 4 lines Fix app_directory when ODBC_STORAGE is being used. The - Makefile did not properly ensure that this information got copied - from what was selected for app_voicemail. (issue #9224) ........ - -2007-03-27 20:11 +0000 [r59272] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Use better english. Renegotiate! Repeat - after me: renegotiate. - -2007-03-27 18:21 +0000 [r59264] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 59261 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59261 | murf | 2007-03-27 12:16:32 -0600 (Tue, 27 Mar 2007) | 1 - line via 9373 (duplicate context in AEL crashes asterisk), - kpfleming pointed on asterisk-dev, that DECLINE in this case the - proper thing to do. This change now has it doing the proper - thing. ........ - -2007-03-27 18:18 +0000 [r59257-59263] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | - 3 lines Fix the check that ensures that the CHANNEL function's - first argument is "rtpqos". Thanks, Corydon. :) ........ - - * /, channels/chan_iax2.c: Merged revisions 59259 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59259 | russell | 2007-03-27 13:05:46 -0500 - (Tue, 27 Mar 2007) | 12 lines Merged revisions 59258 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 - Mar 2007) | 4 lines Fix the use of the "sourceaddress" option - when "bindaddr" is set to 0.0.0.0 instead of having each - interface explicitly listed. (issue #7874, patch by stevens) - ........ ................ - - * /, channels/chan_sip.c, funcs/func_channel.c: Merged revisions - 59256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | - 4 lines Convert the RTPQOS function to just be additional - parameter of the CHANNEL function. This way, it will be possible - for other RTP based channel drivers to expose this information in - the future. ........ - -2007-03-27 14:09 +0000 [r59233-59253] Steve Murphy <murf@digium.com> - - * include/asterisk/config.h: Enhancement via 8118: Realtime API - extension: add methods store_func and destroy_func, to make - Realtime a complete database abstraction - - * pbx/ael/ael-test/ael-test18/extensions.ael (added), - pbx/ael/ael-test/ael-test18 (added), - pbx/ael/ael-test/ref.ael-test18 (added): added the no. 18 - regression test - - * pbx/ael/ael-test/ael-test19/extensions.ael (added), - pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ael-test19 - (added), pbx/ael/ael-test/ref.ael-test7, - pbx/ael/ael-test/ref.ael-test19 (added), - pbx/ael/ael-test/ref.ael-vtest13: updated the regressions with - regards to 9373, the crash on double contexts, and brought other - regressions up to date - - * /, pbx/pbx_ael.c: Merged revisions 59228 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59228 | murf | 2007-03-26 15:41:32 -0600 (Mon, 26 Mar 2007) | 1 - line fix for 9373 (duplicate context in AEL crashes asterisk). I - turned a duplicate context from a WARNING to an ERROR. Now you - get a module load failure, and asterisk just exits. That's better - than a crash, right\? ........ - -2007-03-26 21:46 +0000 [r59229-59231] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) - | 2 lines Change this to a single dp function to make oej happy. - ........ - -2007-03-26 20:27 +0000 [r59226] Steve Murphy <murf@digium.com> - - * /, main/config.c: Merged revisions 59225 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59225 | murf | 2007-03-26 14:06:12 -0600 (Mon, 26 Mar 2007) | 1 - line Fix for 9257; by eliminating the globals in main/config.c, - we make it thread-safe, which is a minimum requirement. ........ - -2007-03-26 19:35 +0000 [r59224] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 59223 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar - 2007) | 2 lines Add ability to specify no timeout. This means as - soon as the prompt is done playing it moves on to the next - priority. ........ - -2007-03-26 18:34 +0000 [r59216-59218] Russell Bryant <russell@digium.com> - - * /: Merged revisions 59217 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59217 | russell | 2007-03-26 13:33:50 -0500 (Mon, 26 Mar 2007) | - 4 lines Somehow the code for building the email for voicemail got - out of sync. This change makes a few tweaks to get 1.4 in sync - with trunk. (issue #9301) ........ - - * /, apps/app_meetme.c: Merged revisions 59215 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59215 | russell | 2007-03-26 13:28:29 -0500 (Mon, 26 Mar 2007) | - 3 lines Fix some codec negotiation problems when CallerID support - is not enabled in SLA. (issue #9308, reported by twilson) - ........ - -2007-03-26 18:14 +0000 [r59214] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 59213 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar - 2007) | 2 lines Make SpeechBackground obey the digit timeout - value. ........ - -2007-03-26 17:57 +0000 [r59211] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Merged revisions 59209 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | - 1 line Rename the new dialplan functions to match the variable - name ........ - -2007-03-26 17:56 +0000 [r59210] Steve Murphy <murf@digium.com> - - * /, main/ast_expr2f.c, pbx/ael/ael.flex, main/ast_expr2.fl: Merged - revisions 59206 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59206 | murf | 2007-03-26 11:38:29 -0600 (Mon, 26 Mar 2007) | 1 - line A fix for the flex input files, DONT_COMPILE, and - STANDALONE_AEL ........ - -2007-03-26 17:51 +0000 [r59208] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: - Merged revisions 59207 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | - 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully - functional in some because they get set in sip_hangup. So, there - are common situations where the variables will not be available - in the dialplan at all. So, this patch provides an alternate - method for getting to this information by introducing AUDIORTPQOS - and VIDEORTPQOS dialplan functions. (issue #9370, patch by - Corydon76, with some testing by blitzrage) ........ - -2007-03-26 16:48 +0000 [r59204-59205] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Fix bug in which parameter type we are - passing. This shouldn't be a problem since both types are the - same underneath. - - * channels/chan_zap.c: Small API related SS7 updates. - -2007-03-26 15:59 +0000 [r59203] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, configure, - include/asterisk/autoconfig.h.in, channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, configure.ac, - channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged - revisions 59202 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 - lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo - cancellation modules, make chan_misdn use it. * add a check for - linux/mISDNdsp.h to configure.ac and update the autogenerated - files: 'configure', 'autoconfig.h.in' (the 'configure' script was - not in sync with the latest configure.ac, so the diff is a bit - bigger than expected). ........ - -2007-03-26 15:20 +0000 [r59201] Joshua Colp <jcolp@digium.com> - - * /, pbx/ael/ael_lex.c: Merged revisions 59200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59200 | file | 2007-03-26 11:16:29 -0400 (Mon, 26 Mar 2007) | 2 - lines Have ast_copy_string magically appear in the aelparse - binary! DONT_OPTIMIZE should now work once again. ........ - -2007-03-24 01:42 +0000 [r59191-59196] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, - 23 Mar 2007) | 10 lines Merged revisions 59194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 - lines Only try to handle a response if it has a response code. - (ASA-2007-011) ........ ................ - - * doc/modules.txt: Update modules.txt to new loader. (issue #9358 - reported by eliel) - -2007-03-23 16:17 +0000 [r59190] Steve Murphy <murf@digium.com> - - * /, apps/app_macro.c: Merged revisions 59188 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59188 | murf | 2007-03-23 10:09:01 -0600 (Fri, - 23 Mar 2007) | 9 lines Merged revisions 59186 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 - line Added a few words in the Macro doc strings about the - behavior of macros with hangups (et al.), as per 9337 ........ - ................ - -2007-03-22 23:41 +0000 [r59181-59183] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59182 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) - | 2 lines don't allow string input to overrun the buffer to hold - it (ASA-2007-010) ........ - - * channels/chan_misdn.c, /: Merged revisions 59180 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 - Mar 2007) | 2 lines remove variables that are no longer used - (--enable-dev-mode is good, developers should be using it) - ........ - -2007-03-22 14:48 +0000 [r59146] Steve Murphy <murf@digium.com> - - * utils/Makefile, /: Merged revisions 59145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59145 | murf | 2007-03-22 08:40:53 -0600 (Thu, 22 Mar 2007) | 1 - line The stuff in utils was compiling with -O6 even if - DONT_OPTIMIZE is set in menuconfig. Added the include to fix that - ........ - -2007-03-21 18:10 +0000 [r59080-59090] Joshua Colp <jcolp@digium.com> - - * /, main/http.c: Merged revisions 59089 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59089 | file | 2007-03-21 14:08:57 -0400 (Wed, 21 Mar 2007) | 2 - lines Add svg mimetype for pari. ........ - - * /, res/res_monitor.c: Merged revisions 59087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r59087 | file | 2007-03-21 14:04:58 -0400 (Wed, - 21 Mar 2007) | 10 lines Merged revisions 59086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 - lines Indicate the filename changed when it is changed. (issue - #9311 reported by jsmith) ........ ................ - - * channels/chan_sip.c: Minor tweak. Only queue up an unhold control - frame if we are actually on hold. This would have shown itself - when a call was initially being setup and the SDP data was being - parsed in. - - * /, channels/chan_sip.c: Merged revisions 59081 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 - lines Until we can do media level parsing for sendrecv/etc just - use the first value found. This crept up when a phone was offered - audio+video and returned an inactive video stream. chan_sip - thought the phone said to put the person on hold but that was - totally wrong. (issue #9319 reported by benbrown) ........ - - * main/db.c: Make the database show command spit out how many - results it got. (issue #9332 reported by junky) - -2007-03-20 21:06 +0000 [r59079] Tilghman Lesher <tlesher@digium.com> - - * /, main/logger.c: Merged revisions 59078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59078 | tilghman | 2007-03-20 16:04:52 -0500 (Tue, 20 Mar 2007) - | 2 lines Fix defines for inline stack backtraces (only used by - developers anyway) ........ - -2007-03-20 20:44 +0000 [r59077] Joshua Colp <jcolp@digium.com> - - * /, channels/iax2-parser.c: Merged revisions 59076 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar - 2007) | 2 lines Copy len variable as well, should fix remaining - IAX2 DTMF issues. ........ - -2007-03-20 18:18 +0000 [r59071-59073] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c, include/asterisk/ael_structs.h: The fix for the - AEL <<security hole>> (bug 9316) is here... - - * /: blocking 59070... it was just a repair, doesn't need to be - here - - * /: blocking 59069... will commit these changes with separate - patch - -2007-03-19 22:32 +0000 [r59051] Joshua Colp <jcolp@digium.com> - - * main/loader.c: It is possible for mod to become invalid after we - unload it (if it's a dynamic module) so move it around a bit. - -2007-03-19 22:31 +0000 [r59050] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 59049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59049 | tilghman | 2007-03-19 17:29:56 -0500 (Mon, 19 Mar 2007) - | 2 lines Oops, this should have been a %d all along ........ - -2007-03-19 15:43 +0000 [r59041] Tilghman Lesher <tlesher@digium.com> - - * configs/sip_notify.conf.sample, /: Merged revisions 59040 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007) - | 2 lines Fix unescaped semicolon (reported via -dev list) - ........ - -2007-03-18 20:39 +0000 [r59038] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 59037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 - lines Issue #9313, Asterisk crash on SIP return code 0 (reported - by qwerty1979) (ASA-2007-011) ........ - -2007-03-18 16:59 +0000 [r59036] BJ Weschke <bweschke@btwtech.com> - - * /, apps/app_followme.c: Merged revisions 59035 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r59035 | bweschke | 2007-03-18 12:36:44 -0400 (Sun, 18 Mar 2007) - | 3 lines Don't return a non-zero return code if the profile - doesn't exist, to match what the documentation says it already - does. (#9307 Reported by kkiely) ........ - -2007-03-16 16:14 +0000 [r58995] Joshua Colp <jcolp@digium.com> - - * /, apps/app_page.c: Merged revisions 58992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2 - lines Wait for the async thread to exit when hanging up all of - the paged phones under all circumstances. (issue #9181 reported - by PhilSmith) ........ - -2007-03-16 01:43 +0000 [r58954-58958] Russell Bryant <russell@digium.com> - - * /, configs/sla.conf.sample: Merged revisions 58957 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15 - Mar 2007) | 1 line fix a couple SLA documentation references - ........ - - * /, build_tools/prep_tarball: Merged revisions 58953 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58953 | russell | 2007-03-15 20:12:40 -0500 (Thu, 15 - Mar 2007) | 2 lines Add the --pdf option to the usage of rubber - in prep_tarball ........ - -2007-03-16 00:04 +0000 [r58949-58950] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /, doc/ast_appdocs.tex: Merged revisions 58946 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58946 | tilghman | 2007-03-15 18:52:48 -0500 (Thu, 15 Mar 2007) - | 2 lines Refashion dump command to match common syntax and - update the resulting appdocs TeX file ........ - - * main/pbx.c: Fix trunk so that it compiles again - -2007-03-15 23:56 +0000 [r58942-58948] Russell Bryant <russell@digium.com> - - * Makefile, /, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: - Merged revisions 58947 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) | - 3 lines Add configure script checking for GTK2 and some - additional Makefile targets to support gmenuselect ........ - - * /, doc/asterisk.tex: Merged revisions 58941 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58941 | russell | 2007-03-15 18:24:09 -0500 (Thu, 15 Mar 2007) | - 1 line add a link to the rubber homepage ........ - -2007-03-15 22:52 +0000 [r58936-58938] Russell Bryant <russell@digium.com> - - * Makefile, /, doc/asterisk.tex: Merged revisions 58937 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58937 | russell | 2007-03-15 17:51:29 -0500 (Thu, 15 Mar 2007) | - 2 lines Add Asterisk version information to the generated PDF - ........ - - * /, build_tools/prep_tarball: Merged revisions 58935 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58935 | russell | 2007-03-15 17:35:52 -0500 (Thu, 15 - Mar 2007) | 2 lines have prep_tarball attempt to build - asterisk.pdf ........ - -2007-03-15 22:33 +0000 [r58934] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_realtime.c: Merged revisions 58933 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58933 | tilghman | 2007-03-15 17:32:33 -0500 (Thu, 15 - Mar 2007) | 2 lines Function works fine, but the documentation is - backwards. ........ - -2007-03-15 22:29 +0000 [r58932] Russell Bryant <russell@digium.com> - - * doc/manager.txt (removed), doc/misdn.txt (removed), - doc/jitterbuffer.tex (added), /, doc/billing.txt (removed), - doc/extensions.tex (added), doc/queues-with-callback-members.tex - (added), doc/localchannel.txt (removed), doc/cdrdriver.txt - (removed), doc/00README.1st (removed), doc/ajam.tex (added), - doc/manager.tex (added), doc/misdn.tex (added), doc/freetds.txt - (removed), doc/odbcstorage.txt (removed), configure, - doc/model.txt (removed), doc/cygwin.txt (removed), doc/sla.tex, - doc/billing.tex (added), doc/ael.txt (removed), - doc/channelvariables.txt (removed), doc/callingpres.txt - (removed), doc/musiconhold-fpm.txt (removed), - doc/localchannel.tex (added), doc/enum.txt (removed), - doc/cdrdriver.tex (added), build_tools/make_buildopts_h, - doc/security.txt (removed), doc/imapstorage.txt (removed), - doc/PEERING, main/pbx.c, doc/freetds.tex (added), - doc/odbcstorage.tex (added), doc/privacy.txt (removed), - configure.ac, doc/iax.txt (removed), doc/channelvariables.tex - (added), doc/ael.tex (added), doc/enum.tex (added), - doc/security.tex (added), doc/math.txt (removed), Makefile, - doc/imapstorage.tex (added), doc/privacy.tex (added), - doc/realtime.txt (removed), doc/dundi.txt (removed), - doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt - (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), - doc/ast_appdocs.tex (added), doc/realtime.tex (added), - doc/ices.txt (removed), doc/dundi.tex (added), doc/queuelog.txt - (removed), doc/extconfig.txt (removed), doc/radius.txt (removed), - doc/cliprompt.tex (added), doc/chaniax.tex (added), - doc/hardware.txt (removed), doc/mp3.txt (removed), - doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex - (added), doc/configuration.txt (removed), doc/queuelog.tex - (added), doc/asterisk-conf.txt (removed), doc/sla.pdf (removed), - doc/ip-tos.txt (removed), doc/hardware.tex (added), doc/h323.txt - (removed), doc/mp3.tex (added), doc/configuration.tex (added), - doc/asterisk-conf.tex (added), doc/jitterbuffer.txt (removed), - doc/channels.txt (removed), doc/ip-tos.tex (added), - doc/extensions.txt (removed), - doc/queues-with-callback-members.txt (removed), doc/apps.txt - (removed), makeopts.in, doc/ajam.txt (removed): Merged revisions - 58931 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | - 13 lines Merge changes from svn/asterisk/team/russell/LaTeX_docs. - * Convert most of the doc directory into a single LaTeX formatted - document so that we can generate a PDF, HTML, or other formats - from this information. * Add a CLI command to dump the - application documentation into LaTeX format which will only be - include if the configure script is run with --enable-dev-mode. * - The PDF turned out to be close to 1 MB, so it is not included. - However, you can simply run "make asterisk.pdf" to generate it - yourself. We may include it in release tarballs or have - automatically generated ones on the web site, but that has yet to - be decided. ........ - -2007-03-15 18:21 +0000 [r58924] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 58923 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 - lines Don't assume that the pvt structure will still exist after - calling schedule_delivery as it may not. (issue #9278 reported by - fmachado) ........ - -2007-03-14 19:19 +0000 [r58904-58907] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 58906 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | - 4 lines Some people like to put "limitonpeer" instead of - "limitonpeers" in their configuration. While we're at it, support - "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172) - ........ - - * /, doc/sla.tex, doc/sla.pdf: Merged revisions 58902 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14 - Mar 2007) | 2 lines Add a more basic example setup to the - examples section ........ - -2007-03-14 17:01 +0000 [r58900-58901] Olle Johansson <oej@edvina.net> - - * cdr/cdr_radius.c: Correct reference to Radius library THanks - Philippe - Greetings from Lisboa, Portugal - - * /, channels/chan_sip.c: Merged revisions 58848 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, - 13 Mar 2007) | 10 lines Merged revisions 58847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 - lines Issue #9229 - No port in request URI on register to non - default SIP ports (neelakantan) ........ ................ - -2007-03-14 16:40 +0000 [r58895-58898] Russell Bryant <russell@digium.com> - - * /, doc/security.txt: Merged revisions 58897 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58897 | russell | 2007-03-14 11:40:22 -0500 - (Wed, 14 Mar 2007) | 11 lines Merged revisions 58896 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 - Mar 2007) | 3 lines Add a note to the security file that the - Asterisk CLI and log files may contain sensitive information, and - that people should keep this in mind. ........ ................ - - * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions - 58894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | - 8 lines By default, don't attempt to do any CallerID handling at - all with SLA because it is known to not work properly in some - situations. However, add an option to enable it for those that - would like to use it anyway. The short story behind this is that - to properly handle CallerID with SLA, we need the ability to - change the CallerID on an existing call, and we are not ready to - handle that. ........ - -2007-03-14 01:56 +0000 [r58881] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 58880 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58880 | tilghman | 2007-03-13 20:47:08 -0500 (Tue, 13 Mar 2007) - | 3 lines Issue 9162 - pbx_substitute_variables_helper assumes - the buffer is initialized to all zeroes. This fixes a case where - it wasn't. ........ - -2007-03-13 23:20 +0000 [r58866-58873] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 58872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) | - 4 lines Ensure that the blinky lights show that the trunk stopped - ringing when the trunk hangs up before a station has answered it. - (issue #9234, reported by francesco_r) ........ - - * /, configs/sla.conf.sample: Merged revisions 58870 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 - Mar 2007) | 1 line fix the reference to the SLA documentation - ........ - - * cdr/cdr_sqlite3_custom.c (added), build_tools/menuselect-deps.in, - configure, include/asterisk/autoconfig.h.in, - configs/cdr_sqlite3_custom.conf (added), - doc/res_config_sqlite.txt (added), cdr/cdr_sqlite.c, - configs/extconfig.conf.sample, configure.ac, UPGRADE.txt, - CHANGES, makeopts.in, res/res_config_sqlite.c (added), - configs/res_config_sqlite.conf (added): Merge changes from - team/russell/sqlite: * Add new module, cdr_sqlite3_custom which - allows logging custom CDRs into a SQLite3 database. (issue #7149, - alerios) * Add new module, res_config_sqlite, which adds realtime - database configuration support for SQLite version 2. I decided - that this was ok since we didn't have any realtime support for - version 3. If someone ports this to version 3, then version 2 - support can be removed or marked deprecated. (issue #7790, - rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of - cdr_sqlite3_custom. Also, note that there were other modules on - the bug tracker that did not make the cut because they provided - some duplicated functionality. Those are: * cdr_sqlite3 (issue - #6754, moy) * cdr_sqlite3 (issue #8694, bsd) - -2007-03-13 10:14 +0000 [r58822-58846] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58845 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 - lines Don't hangup the call on OK or errors on MESSAGE and INFO - inside of a dialog (like video update requests). ........ - - * /, channels/chan_sip.c: Merged revisions 58843 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 - lines Issue #9251 - Clear From URI from user attributes (tgrman) - ........ - - * channels/chan_h323.c: Change URL to OpenH323 (thanks, Tzafrir!) - -2007-03-12 01:22 +0000 [r58780-58784] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 58783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 - lines Allow RFC2833 compensation to compensate for even stupider - implementations by queueing up the end frame at the start, not - the actual end. (issue #8963 reported by AndrewZ) ........ - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 58779 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 - lines Add matchexterniplocally setting which only substitutes - your externip/externhost setting if it matches the localnet - setting. I know of at least two people who need opposite - settings, so I made it an option! (issue #8821 reported by - kokoskarokoska) ........ - -2007-03-11 21:57 +0000 [r58761] Kevin P. Fleming <kpfleming@digium.com> - - * main/asterisk.c: grammatical errors are bad, mmmkay? - -2007-03-11 16:43 +0000 [r58742] Jason Parker <jparker@digium.com> - - * build_tools/cflags.xml, main/asterisk.c: Add CLI command "marko - show birthday" to show "birthday information" for Mark Spencers - upcoming 30th birthday. To enable, run `make menuselect` and - select the option MARKO_BDAY under Compiler Flags. - -2007-03-10 18:15 +0000 [r58639-58706] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 58705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | - 6 lines Fix a few more places in chan_iax2 where the ast_frame - used for receiving a frame was not properly initialized. - - Interpolating a frame when the jitterbuffer is in use - - decrypting a frame when IAX2 encryption is on - frames in an IAX2 - trunk ........ - - * /, apps/app_meetme.c: Merged revisions 58669 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) | - 2 lines Make the compiler happy and initialize a variable. - ........ - - * /, doc/sla.txt (removed), doc/sla.tex (added), doc/sla.pdf - (added): Merged revisions 58638 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) | - 8 lines Merge some updates to the SLA documentation. I plan to - keep working on this to explain all of the expected behavior with - call handling, configuration details for specific phones, and - other things. However, I got tired of doing it in plain text, so - I switched to using LaTeX. I have included the PDF version. I - haven't been able to get a nice looking plain text version out of - it yet, but I'm not terribly concerned since this is supposed to - be more of the manual, while the plain text sample configuration - file is the reference. ........ - -2007-03-09 21:10 +0000 [r58592-58605] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 58604 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2 - lines Fix spelling of unavailable in voicemail documentation. - (issue #9248 reported by tensai) ........ - - * /, channels/chan_sip.c: Merged revisions 58584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, - 09 Mar 2007) | 10 lines Merged revisions 58579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 - lines If we are unable to lookup the host in a c line we have to - abort, otherwise the previous data is gone and we will - (potentially) have no data when all is said and done. ........ - ................ - -2007-03-08 23:21 +0000 [r58511-58541] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 58512 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | - 5 lines Hang up the channel that put the call on hold in the - event processing thread to avoid a race condition. Also, if the - station originated the call that it is putting on hold, don't - hang up the trunk if it was the only station on the call and it - is hanging up due to hold and not a normal hangup. ........ - - * channels/chan_zap.c, /: Merged revisions 58510 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | - 3 lines Add a missing break statement so that handling the above - event does not incorrectly destroy the channel. (issue #9242, - andrew) ........ - -2007-03-08 21:34 +0000 [r58480] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 58479 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007) - | 2 lines Fix segfault (Issue 9236) ........ - -2007-03-08 20:56 +0000 [r58475] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 58474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) | - 3 lines Refactor hold handling a bit so that it does not require - keeping the call up when a call is put on hold. ........ - -2007-03-08 18:05 +0000 [r58390-58437] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 58436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 - lines Make early SDP seeding even smarter! We have to check - codecs in the make_compatible function too. (issue #9221 reported - by marcelbarbulescu) ........ - - * /, main/dsp.c: Merged revisions 58389 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58389 | file | 2007-03-08 11:07:10 -0500 (Thu, - 08 Mar 2007) | 10 lines Merged revisions 58388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 - lines Only print out debug message if the definition that makes - the variables shows up was actually defined. (issue #9233 - reported by serginuez) ........ ................ - -2007-03-08 13:27 +0000 [r58353-58355] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/http.c: Merged revisions 58354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007) - | 2 lines this change was not needed; fclose() handles closing - the file descriptor already ........ - - * /, apps/app_meetme.c, main/http.c: Merged revisions 58351-58352 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007) - | 2 lines fix two cases where HTTP session file descriptors would - not be closed ........ r58352 | kpfleming | 2007-03-08 08:17:42 - -0500 (Thu, 08 Mar 2007) | 2 lines fix a compiler warning, and - overwriting 'res' value ........ - -2007-03-08 01:06 +0000 [r58304-58321] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /, configure, configure.ac: Merged revisions - 58320 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | - 6 lines If we receive ZT_EVENT_REMOVED, destroy the specified - channel. (issue #7256, tzafrir) Also, update the configure script - to make sure that we don't try to build chan_zap if the installed - version of zaptel does not include ZT_EVENT_REMOVED. ........ - - * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Add the - ability to dynamically specify weights for responses to DUNDi - queries. This can be done using a global variable or a dialplan - function. Using the SHELL() function will allow you to use an - external script to determine what the weight in the response - should be. This can be very useful in load balancing - applications. (inspired by discussions with blitzrage and jsmith - in #asterisk-bugs) - -2007-03-07 20:05 +0000 [r58286] Joshua Colp <jcolp@digium.com> - - * main/loader.c: Make the loader less noisy under valgrind. - -2007-03-07 18:20 +0000 [r58244] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 58243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58243 | russell | 2007-03-07 12:19:19 -0600 - (Wed, 07 Mar 2007) | 17 lines (This bug was reported to me by - Kinsey Moore) Merged revisions 58242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | - 7 lines Fix a problem where the Asterisk channel name could be - that of the wrong IAX2 user for a call. This is because the first - step of choosing this name is to look for an IAX2 peer that - happens to have the same IP/port number that this call is coming - from and assuming that is it. However, this is not always - correct. So, I have made it change this name after authentication - happens since at that point, we have an exact match. ........ - ................ - -2007-03-07 17:55 +0000 [r58241] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c, main/rtp.c: Merged revisions 58240 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 - lines Ensure we have (or should have) at least one matching codec - before attempting early bridge SDP seeding. (issue #9221 reported - by marcelbarbulescu) ........ - -2007-03-07 08:08 +0000 [r58224] Olle Johansson <oej@edvina.net> - - * apps/app_ices.c: Adding reference to ices home page. Anyone that - has tested with ices2 ? - -2007-03-07 01:07 +0000 [r58123-58208] Russell Bryant <russell@digium.com> - - * main/file.c: Add the format of the file that is currently being - played to the verbose message. (issue #9105, junky) - - * main/manager.c, /: Merged revisions 58165 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58165 | russell | 2007-03-06 18:25:19 -0600 - (Tue, 06 Mar 2007) | 12 lines Merged revisions 58164 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 - Mar 2007) | 4 lines If the channels acquired using the manager - Redirect action are not up, then don't attempt to do anything - with them. It could lead to weird behavior, including crashes. - (issue #8977) ........ ................ - - * include/asterisk/utils.h: Add some documentation on the arguments - to the base64 encode/decode functions. (inspired by issue #9215) - - * apps/app_queue.c: Send a manager AgentComplete event when the - agent transfers the call, in addition to where it is already sent - if either side hangs up. (issue #9219, rgollent) In passing, I - put this code in a function so it would not be duplicated a third - time. - -2007-03-06 23:19 +0000 [r58122] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 58121 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, - 06 Mar 2007) | 9 lines Merged revisions 58115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 - line Fix for 9220: Eyebeam cannot renew subscriptions for - presence info. Reason: re-SUBSCRIBE requests don't include Accept - headers, which the rfc says are optional (to put it tersely), (it - uses MAY), and luckily, the sip_pvt struct has the format info - stored, so we simply leave it if the format is set, and the - accept header null. ........ ................ - -2007-03-06 23:01 +0000 [r58101-58120] Russell Bryant <russell@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 58119 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | - 3 lines Clarify the documentation of the dialout and - sendvoicemail options. (issue #9000, caio1982 and serge-v) - ........ - - * codecs/codec_zap.c: Sync codec_zap with the one that is in the - 1.4 branch so that it can actually build here, too. - -2007-03-06 20:45 +0000 [r58054-58055] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, - 06 Mar 2007) | 10 lines Merged revisions 58052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 - lines Change error message to proper message ........ - ................ - - * apps/app_stack.c: Debug control, debug control. - -2007-03-06 18:02 +0000 [r58024-58025] Russell Bryant <russell@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 58023 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 - Mar 2007) | 3 lines Return an error of transmit_response is - called without a session. (issue #9002) ........ - -2007-03-06 08:51 +0000 [r57979-57993] Luigi Rizzo <rizzo@icir.org> - - * main/say.c: move declaration to the beginning of a block - - * apps/app_meetme.c: remove duplicate const - -2007-03-05 20:13 +0000 [r57871-57943] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, CHANGES: Add zap show version CLI command. - This pulls the version/echo canceller in use directly using the - ZT_GETVERSION ioctl. (issue #9094 reported by tootai) - - * /, channels/chan_iax2.c: Merged revisions 57914 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 - lines Since chan_iax2 does not support reception of DTMF with - duration ensure that it is set to 0 on the frame. (issue #8521 - reported by gdhgdh) ........ - - * /, apps/app_meetme.c: Merged revisions 57872 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 - lines Don't create a listen channel and record the conference - unless the option is turned on. (issue #9204 reported by - francesco_r) ........ - - * apps/app_meetme.c: I like it when app_meetme builds under dev - mode, don't you? - - * /, apps/app_voicemail.c: Merged revisions 57870 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57870 | file | 2007-03-05 12:52:03 -0500 (Mon, - 05 Mar 2007) | 10 lines Merged revisions 57869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 - lines Make create_dirpath use our standard for return values. -1 - is failure, 0 is success. (issue #9205 reported by ballares) - ........ ................ - -2007-03-05 15:30 +0000 [r57827] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57826 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon, - 05 Mar 2007) | 9 lines Merged revisions 57825 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 - line Fixed a typo introduced via 9156 (either the gotos or their - doc strings are wrong) ........ ................ - -2007-03-05 04:21 +0000 [r57769-57799] Joshua Colp <jcolp@digium.com> - - * /, main/slinfactory.c: Merged revisions 57798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2 - lines Don't allow a NULL pointer to reach ast_frdup. (issue #9155 - reported by cmaj) ........ - - * configs/extensions.conf.sample: Remove no longer present CLI - commands from sample extensions.conf. (issue #9193 reported by - junky) - - * /, res/res_jabber.c: Merged revisions 57770 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2 - lines Don't reference a potentially NULL pointer. (issue #9199 - reported by klolik) ........ - - * /, main/rtp.c: Merged revisions 57768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 - lines Preserve marker bit when P2P bridging. (issue #9198 - reported by edgreenberg) ........ - -2007-03-03 16:43 +0000 [r57736] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Convert stack apps to use ast_storage channel - structure - -2007-03-03 15:35 +0000 [r57708] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-vtest13: - updated the regression tests - -2007-03-03 14:40 +0000 [r57651-57691] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, include/asterisk/channel.h: Expand datastores to - add the notion of inheritance. This will be needed for the - conversion of IAX2 variables from the current custom method to - ast_storage. - - * /, apps/app_voicemail.c: Merged revisions 57649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57649 | tilghman | 2007-03-03 00:45:00 -0600 - (Sat, 03 Mar 2007) | 10 lines Merged revisions 57648 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 - Mar 2007) | 2 lines Memory leak of a list, if call recording was - abandoned ........ ................ - -2007-03-03 01:11 +0000 [r57621] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/say.c: Merged revisions 57620 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007) - | 1 line submitted patch for Georgian language, issue 9010, - submitted by Alexander Shaduri ........ - -2007-03-03 00:01 +0000 [r57557-57590] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: Add the missing configuration template - to the sample config file. Thanks to Lacy Moore on the - asterisk-users list for pointing out that this was missing! - - * /, configure, configure.ac: Merged revisions 57556 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02 - Mar 2007) | 3 lines Update the check that is used to determine - whether zaptel transcoder support is present. The interface has - changed. ........ - -2007-03-02 18:05 +0000 [r57478-57519] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Don't try to do recursive locking/unlocking when it - isn't supported. - - * /, channels/chan_sip.c: Merged revisions 57477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, - 02 Mar 2007) | 10 lines Merged revisions 57475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 - lines If a SIP message comes in and goes to a method handler that - requires additional values that may not be present then send back - an error. ........ ................ - -2007-03-02 17:03 +0000 [r57476] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57473 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri, - 02 Mar 2007) | 9 lines Merged revisions 57458 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 - line further refinement in wording of goto documentation, as per - 9156, goto not proceeding to next instruction ........ - ................ - -2007-03-02 16:59 +0000 [r57474] Russell Bryant <russell@digium.com> - - * apps/app_dumpchan.c, main/cli.c: Add the channel's Language to - the "show channel" CLI command and the DumpChan application. - (issue #9187, Junky) - -2007-03-02 05:57 +0000 [r57438] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c, utils/ael_main.c: Merged revisions 57426 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1 - line I almost had comma escapes right, but 9184 points out the - problem-- the escape is removed by pbx_config, and pbx_ael should - also, before sending it down into the pbx engine. Also, you have - to insert it back in, if you are generating extensions.conf code - from the AEL. ........ - -2007-03-02 00:22 +0000 [r57365-57397] Russell Bryant <russell@digium.com> - - * /, main/file.c: Merged revisions 57396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) | - 4 lines Return the correct digit that interrupted the stream. - This fixes exiting the Background application when using the m - option. (issue #9176, mjagdis) ........ - - * /, apps/app_meetme.c, doc/sla.txt, include/asterisk/channel.h, - configs/sla.conf.sample: Merged revisions 57364 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | - 16 lines Merge changes from svn/asterisk/team/russell/sla_updates - * Originally, I put in the documentation that only Zap interfaces - would be supported on the trunk side. However, after a discussion - with Qwell, we came up with a way to make IP trunks work as well, - using some things already in Asterisk. So, here it is, this now - officially supports IP trunks. * Update the SLA documentation to - reflect how to setup IP trunks. * Add a section in sla.txt that - describes how to set up an SLA system with voicemail. * Simplify - the way DTMF passthrough is handled in MeetMe. * Fix a bug that - exposed itself when using a Local channel on the trunk side in - SLA. The station's channel needs to be passed to the dial API - when dialing the trunk. * Change a WARNING message to DEBUG in - channel.h. This message is of no use to users. ........ - -2007-03-01 22:23 +0000 [r57319] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 57318 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, - 01 Mar 2007) | 10 lines Merged revisions 57317 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 - lines Don't even attempt to optimize things when a proxy channel - is involved. It will just explode in weird and unexplaineable - ways. (issue #9175 reported by clegall_proformatique) ........ - ................ - -2007-03-01 20:24 +0000 [r57293] Russell Bryant <russell@digium.com> - - * main/channel.c: Constify the list of codec preferences. - -2007-03-01 03:01 +0000 [r57259] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. - -2007-03-01 00:08 +0000 [r57241] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: Minor code cleanup... nothing to write home about. - -2007-02-28 23:02 +0000 [r57204-57209] Russell Bryant <russell@digium.com> - - * /, doc/sla.txt, configs/sla.conf.sample: Merged revisions 57207 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | - 2 lines minor tweaks to the sla docs ........ - - * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions - 57203 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | - 7 lines Merge more changes from - svn/asterisk/team/russell/sla_updates * Add support for private - hold. By setting "hold=private" for a trunk, only the station - that put the call on hold will be able to retrieve it from hold. - Also, by setting "hold=private" for a station, any call that - station puts on hold can only be retrieved by that station. - ........ - -2007-02-28 20:46 +0000 [r57184] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, pbx/pbx_dundi.c, include/asterisk/pbx.h, - pbx/pbx_config.c, apps/app_while.c: Convert the PBX core to use - read/write locks. This yields a nifty performance improvement - when it comes to simultaneous calls going through the dialplan. - Using murf's test the old mutex based core took an average of - 57.3 seconds while the rwlock based core took 31.1 seconds. - That's a nifty 26.2 seconds performance improvement. The other - good part is that if we do need to switch back then we just have - to change the lock/unlock API calls. I converted everywhere that - used to touch the mutex locks directly to use them. - -2007-02-28 19:59 +0000 [r57145-57147] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 57146 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) | - 2 lines Minor formatting change ........ - - * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions - 57144 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | - 6 lines Merge changes from svn/asterisk/team/russell/sla_updates - * Add support for the "barge=no" option for trunks. If this - option is set, then stations will not be able to join in on a - call that is on progress on this trunk. ........ - -2007-02-28 19:30 +0000 [r57140] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57139 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57139 | murf | 2007-02-28 12:23:05 -0700 (Wed, - 28 Feb 2007) | 9 lines Merged revisions 57118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 - line a small documentation update, to reflect reality in the goto - doc strings, as per 9156, Goto does not proceed to next prio if - jump fails ........ ................ - -2007-02-28 19:00 +0000 [r57094] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 57093 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, - 28 Feb 2007) | 10 lines Merged revisions 57092 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 - lines Fix a few more issues with the agent logoff CLI command. - (issue #9123 reported by arbrandes) ........ ................ - -2007-02-28 18:21 +0000 [r57090] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions - 57089 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | - 8 lines Merge current set of changes from - svn/asterisk/team/russell/sla_updates * Add support for station - ring delays. Ring delays can be set globally for a station or for - specific trunks on the station. * Fix a few bugs in existing - code. * Restructure and Reorganize code to improve readability - and maintainability. * Improve formatting of the "sla show - (trunks|stations)" CLI commands. ........ - -2007-02-28 17:56 +0000 [r57054-57056] Joshua Colp <jcolp@digium.com> - - * /: Merged revisions 57055 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57055 | file | 2007-02-28 12:55:03 -0500 (Wed, 28 Feb 2007) | 2 - lines Picky compiler... ........ - - * /, apps/app_speech_utils.c: Merged revisions 57053 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb - 2007) | 2 lines Better handle timeouts when the individual speaks - after everything has been played but before the timeout ends. - ........ - -2007-02-28 17:22 +0000 [r57050] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 57049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r57049 | murf | 2007-02-28 10:15:27 -0700 (Wed, 28 Feb 2007) | 1 - line I was surprised that I had not yet downgraded missing goto - targets and macro call defs to a warning, in case they are in - extensions.conf; I rectified this problem. Also, A goto in a - macro to a target in a catch block was not being found; I fixed - this too; the cause was that I needed to treat catch statements - like an extension in the find_match code. ........ - -2007-02-27 22:17 +0000 [r57011] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Properly hangup the original dialed channel, not - the new channel that appeared from the forwarding. (issue #9161 - reported by PhilSmith) - -2007-02-27 17:38 +0000 [r56976] Russell Bryant <russell@digium.com> - - * /: (also issue #9159) Merged revisions 56975 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56975 | russell | 2007-02-27 11:36:09 -0600 (Tue, 27 Feb 2007) | - 4 lines Fix voicemail email attachments. I missed the conversion - of one of the line endings and there was an extra one where it - should not have been. (issue #9128) ........ - -2007-02-27 00:11 +0000 [r56926-56952] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: Issue 7789 - - some telcos want the TON set based on the number, but without the - NANP prefix removed - -2007-02-26 20:43 +0000 [r56889] Russell Bryant <russell@digium.com> - - * /, channels/chan_alsa.c: Merged revisions 56888 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | - 4 lines Restore the behavior of Asterisk 1.2 where if a device - was not specified in alsa.conf, then we just use the system - default, instead of creating our own default of hw:0,0. (issue - #9139) ........ - -2007-02-26 20:09 +0000 [r56860] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 56856 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56856 | file | 2007-02-26 15:07:18 -0500 (Mon, - 26 Feb 2007) | 10 lines Merged revisions 56850 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 - lines Obey the clearglobalvars option in extensions reload (or - dialplan reload depending on your version). (issue #9146 reported - by ramonpeek) ........ ................ - -2007-02-26 20:04 +0000 [r56849] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 56847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | - 2 lines Fix a crash in my last change to iax2_indicate(). (issue - #9150) ........ - -2007-02-26 19:34 +0000 [r56811-56840] Joshua Colp <jcolp@digium.com> - - * /, apps/app_record.c: Merged revisions 56839 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56839 | file | 2007-02-26 14:33:48 -0500 (Mon, 26 Feb 2007) | 2 - lines Update app_record documentation to use new CLI command, - core show file formats. (issue #9151 reported by junky) ........ - - * main/pbx.c, /: Merged revisions 56805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56805 | file | 2007-02-26 12:09:53 -0500 (Mon, 26 Feb 2007) | 2 - lines Use ast_strlen_zero to see if the language and/or context - argument is not present for Background instead of just checking - if it is NULL. (issue #9141 reported by mjagdis) ........ - -2007-02-26 16:54 +0000 [r56786] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 56785 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | - 3 lines Do more complete locking of the chan_iax2_pvt struct in - the indicate callback. (Problem brought up by Ben Smithurst on - the asterisk-dev list) ........ - -2007-02-26 16:38 +0000 [r56784] Joshua Colp <jcolp@digium.com> - - * /, main/asterisk.c: Merged revisions 56783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56783 | file | 2007-02-26 11:36:08 -0500 (Mon, 26 Feb 2007) | 2 - lines Allow both of the show version files and core show file - versions CLI commands to work. (issue #9135 reported by mvanbaak) - ........ - -2007-02-26 01:05 +0000 [r56731-56742] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 56740 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56740 | russell | 2007-02-25 19:04:40 -0600 (Sun, 25 Feb 2007) | - 2 lines Move a comment to be in the correct struct. ........ - - * main/asterisk.c: Remove redundant check to ensure that LOW_MEMORY - is not defined. (issue #9136, mvanbaak) - - * channels/chan_iax2.c: There is no need to look in the iaxs array - for the pvt struct when we already have a pointer to it. - -2007-02-25 14:53 +0000 [r56686] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 56685 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600 - (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 - Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the - channel list, then evaluating additional conditions (e.g. name - prefix) will cause a NULL dereference. ........ ................ - -2007-02-24 20:29 +0000 [r56623-56665] Olle Johansson <oej@edvina.net> - - * include/asterisk/http.h, main/channel.c, - include/asterisk/doxyref.h, include/asterisk/utils.h, - include/asterisk/zapata.h, apps/app_meetme.c, res/res_limit.c, - include/asterisk/config.h, channels/chan_h323.c, pbx/pbx_ael.c, - apps/app_amd.c, include/asterisk/ael_structs.h, - include/asterisk/jingle.h, main/config.c, main/rtp.c: Doxygen - additions, corrections - - * include/asterisk/doxyref.h, channels/chan_zap.c, main/manager.c, - include/asterisk/frame.h: Doxygen updates and corrections - - * apps/app_osplookup.c, funcs/func_curl.c, res/res_snmp.c, - apps/app_festival.c, cdr/cdr_sqlite.c, codecs/codec_speex.c, - contrib/asterisk-ng-doxygen, include/asterisk/jabber.h, - res/res_crypto.c, channels/chan_h323.c, cdr/cdr_pgsql.c, - cdr/cdr_radius.c, apps/app_voicemail.c: Creating new doxygen - macro "\extref" to create page that lists external libraries and - URLs to these. Please help me add these references. We might want - to create a similar macro "\linuxpackage" to list the needed - Linux packages in popular distributions. - - * include/asterisk/jabber.h: Add some external references - - * include/asterisk/doxyref.h, include/asterisk/jabber.h: Doxygen - updates for AJI - The Asterisk Jabber API - -2007-02-24 02:23 +0000 [r56574-56594] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Allow a - Skinny device to monitor a dialplan hint (w00t!). See - skinny.conf.sample for configuration example. Note: Some devices - (seen on 12SP+/30VIP) will lock up if they monitor too many - hints. This seems to be a hardware limitation - there isn't - anything we can do about it. - - * channels/chan_skinny.c: Support devicestate requests. Now you - should be able to subscribe to a Skinny device/line. - - * /, channels/chan_skinny.c: Merged revisions 56569 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb - 2007) | 4 lines Make sure to set a speeddials parent on creation. - Don't crash if hold is pressed when no call is active. Don't - return in places that we shouldn't.. Update softkey map when call - is connected ........ - -2007-02-24 01:56 +0000 [r56564] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Make Meetme build again under dev mode. - -2007-02-23 23:25 +0000 [r56487-56506] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: Merged revisions 56505 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56505 | russell | 2007-02-23 17:24:18 -0600 - (Fri, 23 Feb 2007) | 16 lines Merged revisions 56504 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 - Feb 2007) | 8 lines Fix up a couple more signal handlers to not - do bad things that could cause various undesirable results. The - other day, I made Asterisk deadlock by hitting Control-C because - of a bad signal handler. Now, signal handlers just set a flag and - write to an alert pipe for the flag to be handled. Then, there is - another thread that is monitoring for these flags. If being run - in console mode, it is just the main thread. If Asterisk is in - the background, a thread is created to do it. ........ - ................ - - * channels/chan_iax2.c: Make the hashing function calculate - something that makes more sense. (Thanks to bmd on #asterisk-dev - for pointing out my pointless math). - -2007-02-23 21:57 +0000 [r56458] Joshua Colp <jcolp@digium.com> - - * /, main/sched.c: Merged revisions 56457 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2 - lines Change log notice to debug. It is possible for a scheduled - item to execute and be deleted at close to the same time and - unavoidable. If this happens this message creeps up. ........ - -2007-02-23 21:20 +0000 [r56408-56447] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Merge team/russell/iax2_performance. There - is not a large amount of code here and the changes are not very - invasive. However, they should significantly improve performance - of chan_iax2 under load. IAX2 media frames only carry the - *source* call number. So, when one arrives, the correct session - that it is a part of has to be matched on IP address, port - number, and call number, instead of just a call number. Had these - frames carried the *destination* call number, this would not be - an issue, because that would be a unique identifier that would - make it easy to immediately identify the correct session. The way - that chan_iax2 did this matching was extremely inefficient. It - starts at the first available call number and traverses each call - number sequentially, locking and unlocking a mutex for each one, - to try to match against it. It would do this regardless of - whether the call number was in use or not. So, for a call with a - local call number of 25000, every single incoming media frame - would require a traversal that required 25000 mutex lock and - unlock operations. (Note that the max call number is about 32k). - I have introduced a hash table of active IAX2 calls to improve - this lookup process. The hash is done on the IP address, port - number, and call number. So, for the previous example, a few - lock/unlock operations may be done versus 25000 for each frame. - - * CHANGES: Note that the entries in the CHANGES file only list - functionality changes - - * CHANGES: Add GetConfigJSON to the CHANGES file. - - * /, channels/chan_iax2.c: Merged revisions 56407 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56407 | russell | 2007-02-23 14:20:00 -0600 - (Fri, 23 Feb 2007) | 12 lines Merged revisions 56406 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 - Feb 2007) | 4 lines Don't destroy mutexes before unregistering - all of the entry points from the core. Also, fix a potential - memory leak from not destroying the locks for all of the possible - call numbers (about 32k of them). ........ ................ - -2007-02-23 19:00 +0000 [r56373] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/make_version_h: Merged revisions 56372 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56372 | kpfleming | 2007-02-23 12:59:09 -0600 (Fri, 23 Feb 2007) - | 2 lines build special version strings for AADK/S800i builds - ........ - -2007-02-23 18:01 +0000 [r56278-56342] Russell Bryant <russell@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 56341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) | - 8 lines The IMAP storage code uses the same code to build the - email that is used when voicemail is sent via email using - something like sendmail. In the patch from bug 8033 to fix - various IMAP storage problems, the line endings in the email file - were changed in the code from "\n" to "\r\n". However, this - breaks sending regular voicemail to email. So, this change - conditionally sets line endings to "\r\n" only if IMAP_STORAGE is - enabled. (issue #9128, patch by jarjarbinks, modified by me to - not break IMAP storage) ........ - - * main/manager.c: Introduce a new manager action, GetConfigJSON, - which is intended to improve performance of the GUI. This encodes - the configuration into the JSON format in a manager header, - "JSON: ". The encoded information can be directly used as a - javascript object, so no parsing is needed. For large - configuration files, this can greatly improve loading times in - the GUI. Furthermore, the encoding takes up a lot less space when - being transmitted than the other alternatives. (Inspired by - discussion with Pari) Here is an example of what you get: - http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf - Response: Success JSON: - {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]} - - * main/dial.c, /, apps/app_meetme.c, doc/sla.txt, - configs/sla.conf.sample: Merged revisions 56277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | - 18 lines Merge changes from team/russell/sla_updates. This batch - of changes to the SLA code does a few different things. * I made - the SLA code event driven instead of having to act in a lot of - busy loops while dialing things to wait for state changes. This - makes the code more efficient and readable at the same time. * I - have implemented a couple of new features. The first is inbound - trunk ringing timeouts. This is an option that defines how long - to let an incoming call on a trunk to ring. * I have also - implemented ring timeouts for stations. They may be specified for - the entire station, meaning it is how long to let the station - ring before giving up. You can also specify a ring timeout for a - specific trunk on a station. So, you can say that you only want a - specific station to ring 5 seconds if it is line1 ringing, but - otherwise, there is no timeout. ........ - -2007-02-22 18:53 +0000 [r56232] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /, channels/chan_sip.c: Merged revisions 56231 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, - 22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 - lines Only change the original or clone channel if it's the - channel behind the proxy channel, not if it's just a regular - bridged channel. ........ ................ - -2007-02-22 17:36 +0000 [r56209] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/module.h: move the ast_module_info structure - into the special section as well, otherwise when - restore_globals() is called it will lose its pointer to the - ast_module structure that the loader put there - -2007-02-22 16:48 +0000 [r56188] Joshua Colp <jcolp@digium.com> - - * .cleancount: Since I'm a nice guy... let's increment the clean - count since last night's module changes require a rebuild of - everything essentially. - -2007-02-22 16:25 +0000 [r56187] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Fix some compilation problems in - app_voicemail. There was a parenthesis missing in a function - prototype, and "#elifdef" is not a valid preprocessor directive. - (issue #9122, akohlsmith) - -2007-02-22 13:58 +0000 [r56156] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: Update OSP documention for v1.6. - -2007-02-22 10:46 +0000 [r56126] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 56125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2 - lines Move message from verbose to debug ........ - -2007-02-22 02:48 +0000 [r56095] Steve Murphy <murf@digium.com> - - * /, sounds/Makefile: Merged revisions 56094 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56094 | murf | 2007-02-21 19:39:58 -0700 (Wed, 21 Feb 2007) | 1 - line updated the sound tarball versions in Makefile ........ - -2007-02-22 02:36 +0000 [r56092] Kevin P. Fleming <kpfleming@digium.com> - - * funcs, codecs, apps, include/asterisk/module.h, - Makefile.moddir_rules, Makefile.rules, - build_tools/make_linker_eo_script (added), cdr, pbx, res, - channels, formats, main/loader.c: give embedded modules a helping - hand by backing up and restoring their global variables when they - are loaded and unloaded - -2007-02-22 01:26 +0000 [r56012-56056] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 56055 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) | - 3 lines Restructure a little bit of code to reduce nesting. There - is no functionality change here. ........ - - * /, channels/chan_sip.c: Merged revisions 56011 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r56011 | russell | 2007-02-21 18:57:36 -0600 - (Wed, 21 Feb 2007) | 11 lines Merged revisions 56010 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 - Feb 2007) | 3 lines If we receive a frame that is not in any of - the negotiated formats, then drop it. (potentially issue #8781 - and SPD-12) ........ ................ - -2007-02-22 00:38 +0000 [r56009] Joshua Colp <jcolp@digium.com> - - * /, main/cli.c: Merged revisions 56008 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r56008 | file | 2007-02-21 19:35:55 -0500 (Wed, 21 Feb 2007) | 2 - lines Print out deprecation notice on usage output of CLI - commands. (issue #8925 reported by blitzrage) ........ - -2007-02-22 00:05 +0000 [r55958-56005] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Make filename on email follow subject - message number, purely for cosmetic purposes for individuals like - *cough* jsmith *cough*. (issue #9111 reported by sshah) - - * /, apps/app_meetme.c: Merged revisions 55957 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55957 | file | 2007-02-21 15:35:40 -0500 (Wed, - 21 Feb 2007) | 10 lines Merged revisions 55956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 - lines Change naughty warning message to provide useful - information. If a write now fails on a channel in meetme it will - tell you the channel name instead of spitting out the wrong error - message. ........ ................ - -2007-02-21 20:30 +0000 [r55955] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 55954 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 - lines Fix locking issue, and accept "transport-accept" as a valid - accept message. This should solve issues 8970 and 8503. ........ - -2007-02-21 20:26 +0000 [r55953] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Clarify in the doxygen docs abou RFC2833 - compensation flag. - -2007-02-21 20:23 +0000 [r55952] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 55951 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55951 | russell | 2007-02-21 14:22:33 -0600 (Wed, 21 Feb 2007) | - 3 lines Simplify the last change to app_meetme, and move the call - to dispose_conf() up into the block where we know a conf exists. - ........ - -2007-02-21 20:18 +0000 [r55915-55950] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 55949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55949 | file | 2007-02-21 15:16:34 -0500 (Wed, 21 Feb 2007) | 2 - lines Only dispose of the conference if one was created. ........ - - * /, apps/app_speech_utils.c: Merged revisions 55947 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb - 2007) | 2 lines Only start playing the next file if we have not - been quieted. ........ - - * /, channels/chan_sip.c: Merged revisions 55914 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 - lines Add a flag that indicates whether a SIP dialog is an - outgoing call or not. SIP_OUTGOING originally did it but it was - repurposed to the direction of the last transaction, which can - cause update_call_counter to falsely decrease the wrong counters. - (please don't hurt me oej) (issue #8943 reported by mdu113) - ........ - -2007-02-21 14:07 +0000 [r55870] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/make_version: Merged revisions 55869 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55869 | kpfleming | 2007-02-21 08:06:47 -0600 - (Wed, 21 Feb 2007) | 10 lines Merged revisions 55868 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 - Feb 2007) | 2 lines use new tag version script ........ - ................ - -2007-02-21 08:39 +0000 [r55835] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 55834 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2 - lines Issue #8848 - Turn off lamp more quickly after transfer - (decrement inuse early on transferer's call leg) ........ - -2007-02-21 02:04 +0000 [r55805] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged - revisions 55799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 - lines Fix segfault when buddy couldn't be found. Issue 7764, - patch by sailer ........ - -2007-02-21 01:05 +0000 [r55763] Joshua Colp <jcolp@digium.com> - - * main/dns.c: Return trunk to a state where it compiles under - Darwin. The byte order stuff is ugly, if anyone wants to clean it - up... by all means do so. - -2007-02-21 01:05 +0000 [r55762] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 55758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55758 | russell | 2007-02-20 19:03:25 -0600 (Tue, 20 Feb 2007) | - 4 lines Improve the reference counting to fix bugs where people - report seeing conferences listed that have no members. (issue - #9073) ........ - -2007-02-21 00:14 +0000 [r55671-55748] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 55741 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55741 | file | 2007-02-20 19:11:20 -0500 (Tue, 20 Feb 2007) | 2 - lines Better handle dropped IMAP connections. (issue #9054 - reported by bsmithurst) ........ - - * /, channels/chan_sip.c: Merged revisions 55717 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2 - lines Return behavior I removed. I did not remember that you - could just add a localnet entry to make it work. ........ - - * main/logger.c: Flush out the file pointer. (issue #9115 reported - by guthrie) - - * /, channels/chan_sip.c: Merged revisions 55688 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2 - lines Don't test our own address against the localnet settings. - At least one person has had issues as a result of this from #7051 - so I'm reversing it. (issue #8821 reported by kokoskarokoska) - ........ - - * /, channels/chan_agent.c: Merged revisions 55670 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55670 | file | 2007-02-20 17:47:00 -0500 (Tue, - 20 Feb 2007) | 10 lines Merged revisions 55669 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 - lines Defer clearing callback information if channels are up - until they are hung up. This ensures the hangup process goes - smoothly and no channels get hung in limbo. (issue #8088 reported - by kebl0155) ........ ................ - -2007-02-20 20:32 +0000 [r55591-55635] Russell Bryant <russell@digium.com> - - * /, main/http.c: Merged revisions 55634 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55634 | russell | 2007-02-20 14:26:06 -0600 (Tue, 20 Feb 2007) | - 3 lines Add the Asterisk version information to the Server header - in HTTP responses. (requested by Pari) ........ - - * /, include/asterisk/manager.h: Merged revisions 55590 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) | - 2 lines Increase the maximum number of manager headers to 128, at - the request of Pari. ........ - -2007-02-20 16:56 +0000 [r55556] Jason Parker <jparker@digium.com> - - * channels/chan_jingle.c, /, channels/chan_gtalk.c, - res/res_jabber.c: Merged revisions 55555 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 - lines No need to cast nor free with strdupa (thanks file) 55555! - ........ - -2007-02-20 16:42 +0000 [r55554] Russell Bryant <russell@digium.com> - - * /, configs/sla.conf.sample: Merged revisions 55553 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r55553 | russell | 2007-02-20 10:41:57 -0600 (Tue, 20 - Feb 2007) | 3 lines Change the formatting of sla.conf.sample to - make it more readable. (issue #9112, blitzrage) ........ - -2007-02-20 15:19 +0000 [r55534] Joshua Colp <jcolp@digium.com> - - * res/res_jabber.c: I like it when trunk builds, so let's make - res_jabber compile again! - -2007-02-20 07:48 +0000 [r55514] Olle Johansson <oej@edvina.net> - - * /, res/res_jabber.c: Merged revisions 55483 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55483 | oej | 2007-02-19 22:12:55 +0100 (Mon, 19 Feb 2007) | 3 - lines - Not sending arguments to an application is not "out of - memory" - Making error messages a bit more clear ........ - -2007-02-19 23:27 +0000 [r55495] Jason Parker <jparker@digium.com> - - * .cleancount: We need to bump the cleancount when we make API - changes... - -2007-02-19 18:15 +0000 [r55436] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 55435 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55435 | tilghman | 2007-02-19 12:11:48 -0600 - (Mon, 19 Feb 2007) | 10 lines Merged revisions 55434 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 - Feb 2007) | 2 lines forcename and forcegreetings options should - check to see if the recording already exists ........ - ................ - -2007-02-19 16:01 +0000 [r55410-55414] Joshua Colp <jcolp@digium.com> - - * CHANGES: Clarify last change for SMDI in CHANGES file. - - * configs/voicemail.conf.sample, apps/app_voicemail.c: Allow both - an external application and SMDI to do voicemail notification at - the same time. (issue #8625 reported by lters) - -2007-02-19 15:24 +0000 [r55409] Doug Bailey <dbailey@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 55397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | - 3 lines Changed iax2 process thread to detached to correct memory - leak due to left over thread context on thread exit. Modified - module unload process to avoid deadlocks on pthread cancels - ........ - -2007-02-18 22:07 +0000 [r55375] Olle Johansson <oej@edvina.net> - - * apps/app_voicemail.c: Formatting changes. - -2007-02-18 19:13 +0000 [r55351-55352] Joshua Colp <jcolp@digium.com> - - * codecs/gsm/inc/proto.h: Return GSM to a state where it actually - builds under dev mode. - - * channels/chan_h323.c: Update chan_h323 to new set_rtp_peer - definition. - -2007-02-18 15:11 +0000 [r55330] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Being picky... - -2007-02-18 15:03 +0000 [r55329] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, channels/chan_misdn.c, main/srv.c, - main/editline/refresh.c, pbx/ael/ael.tab.c, - channels/misdn/isdn_msg_parser.c, channels/chan_oss.c, - main/enum.c, apps/app_voicemail.c, main/ast_expr2.c: add -Wundef - to the --enable-dev-mode flags, so that mistyped macro names in - #if expressions will be caught convert various #if expressions to - #ifdef for macros that may not be defined (and where the value is - not important) Note: two of these changes are in bison generated - files which is going to be inconvenient when they are regenerated - -2007-02-18 15:01 +0000 [r55279-55323] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Simplify post_manager_event() - - * /, apps/app_record.c: Merged revisions 55278 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55278 | oej | 2007-02-18 13:35:54 +0100 (Sun, - 18 Feb 2007) | 10 lines Merged revisions 55277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 - lines Documentation update (#9053, jsmith) ........ - ................ - -2007-02-17 17:41 +0000 [r55220] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 55219 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55219 | file | 2007-02-17 12:39:32 -0500 (Sat, 17 Feb 2007) | 2 - lines Add missing membername option to AddQueueMember - documentation. (issue #9088 reported by seanbright) ........ - -2007-02-17 17:11 +0000 [r55218] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 55217 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb - 2007) | 4 lines Fix an issue where callerid would not be - displayed on some phones. Issue 8995, initial patch and research - done by wedhorn ........ - -2007-02-17 16:48 +0000 [r55087-55198] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: We want to skip the queue if the name doesn't - match the specified one, not if they *do*. - - * apps/app_queue.c: Increase "queue show" buffer size from 80 to - 240. This should be more then enough for most cases. (issue #9089 - reported by mvanbaak) - - * apps/app_dial.c, /: Merged revisions 55154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55154 | file | 2007-02-16 22:55:30 -0500 (Fri, - 16 Feb 2007) | 10 lines Merged revisions 55153 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 - lines Answer the channel before recording privacy information. - (issue #8926 reported by lmamane) ........ ................ - - * /, apps/app_queue.c: Merged revisions 55129 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55129 | file | 2007-02-16 21:59:50 -0500 (Fri, 16 Feb 2007) | 2 - lines Make the 'i' option of Queue actually work. (issue #8986 - reported by utis) ........ - - * channels/chan_jingle.c: Update chan_jingle to new definition of - set_rtp_peer. - - * /, channels/chan_sip.c: Merged revisions 55086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, - 16 Feb 2007) | 10 lines Merged revisions 55073 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 - lines Allow chan_sip to handle attended transfers from a SIP - phone that is sitting behind chan_agent. Yes folks, all it took - was one line of code. (issue #8784 reported by pzieba) ........ - ................ - -2007-02-17 01:11 +0000 [r55004-55077] Russell Bryant <russell@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 55052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) | - 3 lines If the pg_config application is found, but there is - probably executing it, then consider postgres unavailable. (issue - #8637) ........ - - * /, codecs/gsm/Makefile: Merged revisions 55050 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r55050 | russell | 2007-02-16 18:31:42 -0600 (Fri, 16 Feb 2007) | - 3 lines Filter out yet another architecture that does not work - with the optimizations in the built-in libgsm. (issue 8637, ovi) - ........ - - * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged - revisions 55006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55006 | russell | 2007-02-16 16:49:42 -0600 - (Fri, 16 Feb 2007) | 17 lines Merged revisions 55005 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 - Feb 2007) | 9 lines Revert the change I did in revisions 54955, - 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a - conference is created from meetme.conf, it is acceptable behavior - that the pin can not be changed until the conference goes away. I - also added a note in meetme.conf to describe this behavior. We - still have another issue in 1.4 and trunk where some conferences - with no users don't go away. That is the real bug that needs to - be addressed here. ........ ................ - - * apps/app_dumpchan.c: Print the raw read/write formats in the - DumpChan application. (issue #9083, junky) - -2007-02-16 22:20 +0000 [r55003] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 55002 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, - 16 Feb 2007) | 10 lines Merged revisions 54999 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 - lines Do not send indications through ast_indicate in chan_agent - but instead go directly to the technology. This way when - indications are emulated they happen on the Agent channel and do - not screw up formats on the channels. (issue #8439 reported by - punkgode) ........ ................ - -2007-02-16 21:13 +0000 [r54970] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 54969 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r54969 | russell | 2007-02-16 15:12:18 -0600 - (Fri, 16 Feb 2007) | 13 lines Merged revisions 54955 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 - Feb 2007) | 5 lines For conferences that are configured in - meetme.conf, check the configuration file every time someone - joins the conference instead of only when the conference is first - created. This is to ensure that changes to the pin numbers in the - config file are always honored. (issue #9073) ........ - ................ - -2007-02-16 18:53 +0000 [r54910-54925] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 54924 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2 - lines Need to check macro extension as well as macro context for - directed pickup. ........ - - * res/res_features.c, configs/features.conf.sample: Allow the user - to specify where to enable the respective features for when a - parked call is picked up. (ie: transfers and parking) - -2007-02-16 18:04 +0000 [r54890-54901] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 54898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54898 | russell | 2007-02-16 12:03:41 -0600 (Fri, 16 Feb 2007) | - 4 lines Fix setting "autofallthrough" to yes by default. It was - set to enabled in pbx.c. However, if the option was not present - in extensions.conf, then pbx_config.c would set it back to - disabled. ........ - - * /, res/res_features.c: Merged revisions 54888 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54888 | russell | 2007-02-16 11:40:38 -0600 (Fri, 16 Feb 2007) | - 3 lines Clean up a few coding guidelines issues - spaces to tabs, - use sizeof() to pass the size of a static buffer, add spaces ... - ........ - -2007-02-16 17:41 +0000 [r54889] Joshua Colp <jcolp@digium.com> - - * res/res_features.c, CHANGES, configs/features.conf.sample: Add - option to features.conf that enables parking via DTMF on picked - up parked calls. (issue #9082 reported by francesco_r) - -2007-02-16 17:26 +0000 [r54887] Jason Parker <jparker@digium.com> - - * /, main/asterisk.c: Merged revisions 54886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54886 | qwell | 2007-02-16 11:25:21 -0600 (Fri, 16 Feb 2007) | 4 - lines Clarify a restart message. It's silly, but the reporter had - a very valid point. Issue 9079 ........ - -2007-02-16 17:07 +0000 [r54885] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 54884 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2 - lines Allow directed pickup to pick up the real context instead - of the macro context if a Macro is used. (issue #8984 reported by - jamesb63) ........ - -2007-02-16 14:31 +0000 [r54773-54862] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Formatting, whitespace fixes - - * apps/app_voicemail.c: More cleanups of app_voicemail - - * CREDITS, main/channel.c, channels/chan_sip.c, - channels/chan_skinny.c, include/asterisk/rtp.h, - include/asterisk/channel.h, channels/chan_gtalk.c, CHANGES, - include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c: - Adding Realtime Text support (T.140) to Asterisk T.140/RFC 2793 - is a live communication channel, originally created for IP based - text phones for hearing impaired. Feels very much like the old - Unix talk application. This code is developed and disclaimed by - John Martin of Aupix, UK. Tested for interoperability by myself - and Omnitor in Sweden, the company that wrote most of the - specifications. A big thank you to everyone involved in this. - - * /, channels/chan_sip.c: Merged revisions 54787 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2 - lines Issue #7541 - Handle multipart attachments to SIP messages - - even if boundary is quoted. ........ - - * res/res_agi.c: Issue #9068 - make sure we quote HTML characters - correctly too (seanbright) - - * /, res/res_agi.c: Merged revisions 54772 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r54772 | oej | 2007-02-16 12:39:55 +0100 (Fri, - 16 Feb 2007) | 10 lines Merged revisions 54771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 - lines Issue #9069 - If we open with TH we should not close with - /TD. (seanbright) ........ ................ - -2007-02-16 01:36 +0000 [r54711-54749] Joshua Colp <jcolp@digium.com> - - * main/acl.c: Rely on ast_gethostbyname to handle IP addresses, not - inet_aton. (issue #9056 reported by pj) - - * CHANGES, apps/app_chanspy.c: Add 'o' option to Chanspy which - causes it to only listen to audio coming from the channel, and - the 'X' option which allows the user to exit to a valid single - digit extension. (issue #8137 reported by mnicholson) - - * /, apps/app_speech_utils.c: Merged revisions 54714 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb - 2007) | 2 lines Don't let dtmf leak over into the engine and let - it skew the results... also give DTMF results priority. (issue - #9014 reported by surftek) ........ - - * main/manager.c: Properly handle an error result from a manager - action. This could have left the action list permanently locked - for reading. - -2007-02-15 20:29 +0000 [r54654-54686] Olle Johansson <oej@edvina.net> - - * apps/app_voicemail.c: - add some notes, asking for help - insert - a few ast_strlen_zero - Doxygen additions - A few more spaces - - * main/io.c: Make file's new comment doxygenified - -2007-02-15 16:24 +0000 [r54624] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 54623 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r54623 | file | 2007-02-15 11:19:39 -0500 (Thu, - 15 Feb 2007) | 10 lines Merged revisions 54622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 - lines Use a separate variable to indicate execution should - continue instead of the return value. (issue #8842 reported by - pluto70) ........ ................ - -2007-02-15 15:53 +0000 [r54574-54599] Olle Johansson <oej@edvina.net> - - * CHANGES: ...and don't forget to update CHANGES - - * channels/chan_sip.c: Add callgroup and pickupgroup to SIPPEER - function. (thanks ramon) - - * CHANGES: Update CHANGES - - * channels/chan_sip.c, configs/extconfig.conf.sample, - doc/realtime.txt: Issue #7443 - amdtech - Optionally SIP - registrations in another realtime family. - -2007-02-15 02:11 +0000 [r54489-54552] Joshua Colp <jcolp@digium.com> - - * main/io.c: Clean up the I/O context handler. - - * apps/app_flash.c, apps/app_image.c, apps/app_exec.c: Few more - code clean ups. - - * apps/app_milliwatt.c: Clean up app_milliwatt code. - - * apps/app_dial.c, /: Merged revisions 54481 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2 - lines Forward begin DTMF frames as well as end. (issue #9068 - reported by mhardeman) ........ - -2007-02-14 20:45 +0000 [r54464-54466] Olle Johansson <oej@edvina.net> - - * main/asterisk.c: Show version in "core show settings" - - * CHANGES: Updates and re-organization to make it easier to digest - this information - - * main/cdr.c, main/manager.c, include/asterisk/config.h, - include/asterisk/cdr.h, include/asterisk/manager.h, - main/asterisk.c, main/config.c: New CLI command "Core show - settings" to list some core settings - -2007-02-14 17:14 +0000 [r54404] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 54375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed, - 14 Feb 2007) | 10 lines Merged revisions 54373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 - lines When handling glare on a PRI, move the requested channel - rather than hang up the old one. Fix for 8957 and 9011. ........ - ................ - -2007-02-14 17:02 +0000 [r54348-54379] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Make documentation match the source - code. - - * channels/chan_sip.c: Issue #9060 - host= parameter in sip.conf - stopped working caused by outbound proxy patch. - - * channels/chan_sip.c: Add port number to SIPPEER dialplan function - -2007-02-14 08:34 +0000 [r54325] Paul Cadach <paul@odt.east.telecom.kz> - - * codecs/codec_g722.c: I don't know how it worked earlier, but - valgrind produces core every time you try to load codec_g722. - Fixed. ;-) - -2007-02-14 01:12 +0000 [r54291] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 54290 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 - lines Add G722 to ast_best_codec. If anyone disagrees with it's - placement, feel free to change it. (issue #9045 reported by gork) - ........ - -2007-02-13 22:02 +0000 [r54067-54261] Russell Bryant <russell@digium.com> - - * include/asterisk/devicestate.h, apps/app_meetme.c, - res/res_features.c, include/asterisk/cli.h, main/devicestate.c, - CHANGES, apps/app_queue.c, funcs/func_devstate.c (added), - main/cli.c: This introduces a new dialplan function, DEVSTATE, - which allows you to do some pretty cool things. First, you can - get the device state of anything in the dialplan: NoOp(SIP/mypeer - has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234 - has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows - you to create custom device states so you can control phone lamps - directly from the dialplan. - Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten => - mycustomlamp,hint,Custom:mycustomlamp - - * /, channels/chan_sip.c: Merged revisions 54204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | - 5 lines If we fail to create the SIP socket, then return -1 from - reload_config() so that load_module() will return - AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get - spammed with error messages every time chan_sip tries to send a - message. ........ - - * /, channels/chan_sip.c: Merged revisions 54235 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) | - 2 lines Remove a couple of leftover debug messages ........ - - * include/asterisk/devicestate.h, /: Merged revisions 54218 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) | - 3 lines Fix the documentation on the return values from device - state provider registration and deletion. ........ - - * main/asterisk.c: Use spaces instead of tabs in the help text for - a CLI command - - * main/asterisk.c: Simplify WELCOME_MESSAGE to be a single function - call instead of one for each line. - - * include/asterisk/cli.h, main/asterisk.c, main/cli.c: - Constify - the format string passed to ast_cli() - Simplify printing out the - warranty and license - - * main/dial.c, /, include/asterisk/dial.h: Merged revisions 54103 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | - 2 lines Change ast_set_state_callback() to - ast_dial_set_state_callback() ........ - - * main/dial.c, /, apps/app_meetme.c, apps/app_page.c, - include/asterisk/dial.h: Merged revisions 54066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | - 4 lines - Add the ability to register a callback to monitor state - changes in an asynchronous dial operation. - Rename the various - references to "status" to "state" in the dial API ........ - -2007-02-12 15:48 +0000 [r54003-54004] Russell Bryant <russell@digium.com> - - * configs/users.conf.sample, /: Merged revisions 54002 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12 - Feb 2007) | 2 lines Fix a typo where "vmpassword" should be - "vmsecret" ........ - - * main/channel.c: Simplify a small bit of logic. - -2007-02-12 02:44 +0000 [r53980] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_realtime.c: Formatting fixes - -2007-02-11 20:49 +0000 [r53914-53953] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Be careful with debug messages in trunk, - they tend to stay around for release.... - - * channels/chan_sip.c: Small fix in outbound proxy support. - - * channels/chan_sip.c, configs/sip.conf.sample: Add support for - outbound proxy for peers and [general] This replaces the older, - broken, implementation where a setting in [general] did not do - anything and the [peer] part was broken. - - * main/acl.c: Fix debug handling in acl.c - -2007-02-10 09:23 +0000 [r53882-53885] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/chan_h323.c: Merged revisions 53881 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) | - 1 line Fix VLDTMF reception ........ - - * /, apps/app_echo.c: Merged revisions 53880 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53880 | pcadach | 2007-02-10 01:08:55 -0800 (Сбт, 10 Фев 2007) | - 1 line Much simpler than previous one ;-) ........ - - * main/channel.c, /: Merged revisions 53879 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) | - 1 line Provide correct DTMF duration ........ - -2007-02-10 06:14 +0000 [r53851] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, configure.ac: Merged revisions 53850 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r53850 | kpfleming | 2007-02-10 00:06:08 -0600 (Sat, 10 - Feb 2007) | 3 lines don't display the --with-imap message unless - --with-imap was specified without a path use '-n' instead of '! - -z' for tests ........ - -2007-02-10 00:42 +0000 [r53784-53819] Russell Bryant <russell@digium.com> - - * include/asterisk/app.h, include/asterisk/utils.h, main/dial.c, /, - apps/app_meetme.c, channels/chan_sip.c, doc/sla.txt (added), - include/asterisk/dial.h, configs/sla.conf.sample: Merged - revisions 53810 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | - 24 lines Merge team/russell/sla_rewrite This is a completely new - implementation of the SLA functionality introduced in Asterisk - 1.4. It is now functional and ready for testing. However, I will - be adding some additional features over the next week, as well. - For information on how to set this up, see - configs/sla.conf.sample and doc/sla.txt. In addition to the - changes in app_meetme.c for the SLA implementation itself, this - merge brings in various other changes: chan_sip: - Add the - ability to indicate HOLD state in NOTIFY messages. - Queue HOLD - and UNHOLD control frames even if the channel is not bridged to - another channel. linkedlists.h: - Add support for rwlock based - linked lists. dial.c: - Add the ability to run ast_dial_start() - without a reference channel to inherit information from. ........ - - * channels/chan_jingle.c: add another dependency - - * /, apps/app_echo.c: Merged revisions 53783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) | - 4 lines When the Echo() application receives the digit '#', echo - that back as well. Since we already sent the BEGIN frame for that - digit, it makes sense to send the END as well. ........ - -2007-02-09 23:53 +0000 [r53782] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/get_moduleinfo, res/res_config_odbc.c, /, - build_tools/get_makeopts, funcs/func_odbc.c, res/res_adsi.c, - channels/chan_gtalk.c, apps/app_adsiprog.c, apps/app_voicemail.c: - Merged revisions 53779-53781 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) - | 2 lines fix awk scripts to work when both MODULEINFO and - MAKEOPTS are present in a source file ........ r53780 | kpfleming - | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines add some - inter-module dependencies ........ r53781 | kpfleming | - 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines another - dependency ........ - -2007-02-09 19:39 +0000 [r53717-53750] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 53749 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2 - lines Temporarily change musicclass on channel to one specified - in Dial so that the 'm' option functions properly. (issue #8969 - reported by christianbee) ........ - - * apps/app_queue.c: Clean up documentation of Queue application. - (issue #9022 reported by seanbright) - -2007-02-09 16:43 +0000 [r53716] Kevin P. Fleming <kpfleming@digium.com> - - * doc/imapstorage.txt, /, configure, configure.ac: Merged revisions - 53715 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007) - | 2 lines clarify the fact that voicemail IMAP storage cannot be - built against a distro's binary c-client library package (at - least not at this time) ........ - -2007-02-09 01:57 +0000 [r53602-53691] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: I'm crazy so I think I'll change the - musiconhold classes linked list to read/write as well! - - * main/manager.c: It is with pleasure that I announce the return of - rawman support through the HTTP server. (issue #9013 reported by - Jynger) - - * /, apps/app_speech_utils.c: Merged revisions 53601 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb - 2007) | 2 lines Fix timeout issue when utterance is longer then - timeout itself. ........ - -2007-02-08 17:19 +0000 [r53580] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Rename this instance of "busy limit" to - "busy level" as well - -2007-02-08 16:41 +0000 [r53577] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: rename busy-limit - to busy-level, since it is not a limit actually parse the - busy-limit option from sip.conf, instead of ignoring it - -2007-02-08 13:50 +0000 [r53531-53533] Tilghman Lesher <tlesher@digium.com> - - * /, main/loader.c: Merged revisions 53532 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53532 | tilghman | 2007-02-08 07:47:54 -0600 (Thu, 08 Feb 2007) - | 2 lines Issue 9007 - Mutex not released on early return - ........ - - * /, apps/app_voicemail.c: Merged revisions 53530 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53530 | tilghman | 2007-02-08 07:40:02 -0600 - (Thu, 08 Feb 2007) | 10 lines Merged revisions 53529 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 - Feb 2007) | 2 lines Issue 9003 - If fullname is empty, quote() - passes back "\"" ........ ................ - -2007-02-07 23:56 +0000 [r53465-53498] Russell Bryant <russell@digium.com> - - * /, main/db1-ast/Makefile: Merged revisions 53497 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07 - Feb 2007) | 6 lines When building libdb1.a, put the additional - flags needed at the beginning of ASTCFLAGS, instead of at the - end. This way, we ensure that we find the local headers first - before accidentally trying to use headers that exist in locations - specified in the ASTCFLAGS passed from the main Makefile. (issue - #8637, ovi) ........ - - * /, main/Makefile: Merged revisions 53464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) | - 4 lines The clean target actually needs to run "distclean" on - editline. This is because we need to make sure that its configure - script gets executed again, because the CFLAGS we want to pass to - editline may have changed. ........ - -2007-02-07 17:57 +0000 [r53435] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 53434 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 - lines We can not reliably do P2P bridging with DTMF passing back - with compensation if we need to listen for DTMF frames. (issue - #8962 reported by caio1982) ........ - -2007-02-07 17:46 +0000 [r53431] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: Merged revisions 53429 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | - 7 lines When parsing the NTP timestamp in a sender report - message, you are supposed to take the low 16 bits of the integer - part, and the high 16 bits of the fractional part. However, the - code here was erroneously taking the low 16 bits of the - fractional part. It then shifted the result 16 bits down, so the - result was always zero. This fix makes it grab the appropriate - high 16 bits, instead. (issue #8991, pointed out by - andre_abrantes) ........ - -2007-02-07 17:06 +0000 [r53359-53400] Joshua Colp <jcolp@digium.com> - - * /, apps/app_playback.c: Merged revisions 53399 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53399 | file | 2007-02-07 12:04:44 -0500 (Wed, 07 Feb 2007) | 2 - lines Directly load say.conf in load_module instead of calling - the reload function. (issue #8946 reported by junky) ........ - - * /, channels/chan_iax2.c: Merged revisions 53358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, - 07 Feb 2007) | 10 lines Merged revisions 53357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 - lines Fix a few potential memory leaks with realtime users and - peers. (issue #8999 reported by bsmithurst) ........ - ................ - -2007-02-07 15:35 +0000 [r53356] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_macro.c: Merged revisions 53355 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53355 | tilghman | 2007-02-07 09:33:51 -0600 - (Wed, 07 Feb 2007) | 10 lines Merged revisions 53354 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 - Feb 2007) | 2 lines Issue 7440 - Macro called from Macro from the - h extension exits prematurely ........ ................ - -2007-02-07 09:51 +0000 [r53334] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 53324 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53324 | crichter | 2007-02-07 10:22:44 +0100 - (Mi, 07 Feb 2007) | 9 lines Merged revisions 52843 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 - Jan 2007) | 1 line fixed some possible segfaults. also fixed an - very important bug which occurs on high load (when calls are very - fast generated) ........ ................ - -2007-02-07 05:25 +0000 [r53247-53297] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_jabber.c: Merged revisions 53294 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53294 | tilghman | 2007-02-06 23:24:31 -0600 (Tue, 06 Feb 2007) - | 2 lines Text fix for jabber reload command (reported by bkruse - via IRC) ........ - - * main/manager.c, /: Merged revisions 53246 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53246 | tilghman | 2007-02-06 01:00:52 -0600 - (Tue, 06 Feb 2007) | 10 lines Merged revisions 53245 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 - Feb 2007) | 2 lines Issue 8987 - Status could return two - responses (mnicholson) ........ ................ - -2007-02-05 21:55 +0000 [r53200] Olle Johansson <oej@edvina.net> - - * main/io.c: Doxygen formatting changes - -2007-02-05 17:06 +0000 [r53151-53153] Joshua Colp <jcolp@digium.com> - - * /, apps/app_playback.c: Merged revisions 53152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2 - lines Ensure say_cfg is NULL when the module is loaded. (issue - #8946 reported by junky) ........ - - * /, apps/app_playback.c: Merged revisions 53150 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2 - lines Unregister Playback CLI commands as well as dialplan - application. (issue #8946 reported by junky) ........ - -2007-02-05 00:30 +0000 [r53144] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 53143 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 - lines Add some comments on queue system behaviour and how it - affects the SIP channel ........ - -2007-02-03 22:06 +0000 [r53140-53142] Tilghman Lesher <tlesher@digium.com> - - * UPGRADE.txt: Deprecate SetCallerPres application - - * apps/app_setcallerid.c, funcs/func_callerid.c: Add CALLERPRES - dialplan function and deprecate SetCallerPres application - - * funcs/func_odbc.c: Fix compiler warnings - -2007-02-03 21:06 +0000 [r53139] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 53138 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 - lines Make SIPDtmfMode application work with recent capability - changes, and also fix an RTP stack issue when the auto option was - used. (issue #8972 reported by mdu113) ........ - -2007-02-03 20:46 +0000 [r53137] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, /: Merged revisions 53136 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53136 | russell | 2007-02-03 14:44:20 -0600 - (Sat, 03 Feb 2007) | 12 lines Merged revisions 53133 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 - Feb 2007) | 4 lines set the DIALSTATUS variable to contain - "INVALIDARGS" when the dial application exits early because of - invalid arguments instead of just leaving it empty. (issue #8975) - ........ ................ - -2007-02-03 10:12 +0000 [r53132] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx: Merged revisions 53131 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) | - 1 line Remove quote from H.323 vendor string because due to - compatibilities with Nortel Meridian CS1000 reported at - www.voip-info.org ........ - -2007-02-02 20:05 +0000 [r53126-53127] Olle Johansson <oej@edvina.net> - - * doc/queue.txt: Update with info about SIP channels and queues - - * doc/queue.txt (added): Adding a template for documentation on - call queues. Please help us add to this! Thanks /OEJ and BJ - -2007-02-02 18:21 +0000 [r53111-53125] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Add onHold value to sip show inuse as well. - - * /, main/rtp.c: Merged revisions 53120 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 - lines Correct a copy/pasted error message line for RTCP. ........ - - * /, main/config.c: Merged revisions 53118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53118 | file | 2007-02-02 10:59:53 -0600 (Fri, - 02 Feb 2007) | 10 lines Merged revisions 53117 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 - lines Pass the glob expanded filename to process_text_line so - that error messages contain the actual filename, not the original - include one. (issue #8959 reported by tzafrir) ........ - ................ - - * Makefile, /: Merged revisions 53114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53114 | file | 2007-02-02 09:29:35 -0600 (Fri, 02 Feb 2007) | 2 - lines Add systemname to asterisk.conf generation per recent - discussions about it. (issue #8968 reported by blitzrage) - ........ - - * main/devicestate.c: Clean up ast_device_state. It's pretty now! - - * main/devicestate.c: Switch the devicestate thread to operate the - same way as the logging thread. Pops all entries off the list to - be processed, resets the list back to a clean state, and - processes each entry. The thread won't have to acquire the list - lock again until it checks to see if there are more to process. - - * main/devicestate.c: Read/write lockify the devicestate stuff a - bit. - -2007-02-02 00:26 +0000 [r53110] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Patch based on - this patch with small changes for trunk... Merged revisions 53109 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 - lines Disable the direct p2p RTP call setup in SIP. You can - enable it in sip.conf, but it is now considered experimental - until we solve the AST_CONTROL_ANSWER with payload and videocaps - stuff. ........ - -2007-02-01 22:26 +0000 [r53098-53105] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 53104 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, - 01 Feb 2007) | 10 lines Merged revisions 53103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 - lines Copy noncodeccapability over to the joint variable so that - telephone-event will get transmitted in the sent INVITE. ........ - ................ - - * /, channels/chan_sip.c: Merged revisions 53097 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, - 01 Feb 2007) | 10 lines Merged revisions 53095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 - lines Don't negotiate RFC2833 when not configured to do so. - (issue #8799 reported by mdu113) ........ ................ - -2007-02-01 21:27 +0000 [r53094] Russell Bryant <russell@digium.com> - - * /, funcs/func_strings.c: Merged revisions 53093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53093 | russell | 2007-02-01 15:24:52 -0600 (Thu, 01 Feb 2007) | - 2 lines Fix the FIELDQTY function to not crash. (reported by - blitzrage and Corydon on IRC) ........ - -2007-02-01 21:17 +0000 [r53092] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 53085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 - lines - Clean INC_COUNT flag when we decrement call counter - If - it's still set at time of dialog destruction, make sure we - decrement the device call counter properly before we destroy the - dialog ........ - -2007-02-01 21:12 +0000 [r53087-53089] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 53088 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53088 | file | 2007-02-01 15:11:28 -0600 (Thu, - 01 Feb 2007) | 10 lines Merged revisions 53084 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2 - lines Return previous behavior of having MOH pick up where it was - left off. (issue #8672 reported by sinistermidget) ........ - ................ - -2007-02-01 20:44 +0000 [r53080-53083] Olle Johansson <oej@edvina.net> - - * /, apps/app_queue.c: Merged revisions 53081 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53081 | oej | 2007-02-01 21:38:58 +0100 (Thu, 01 Feb 2007) | 2 - lines Change debug level for state change message that is not - really informative when debugging app_queue ........ - - * channels/chan_sip.c, configs/sip.conf.sample: Implementing - "busy-limit". If you set call limit and busy limit, chan_sip will - indicate BUSY for a device that has reached the busy limit and - allow calls up to the call limit, allowing for call transfers - (that generate a new call). If you only set call limit, chan_sip - will not indicate BUSY until that limit is filled. This affects - SIP subscriptions, call queues and manager applications. - - * /, channels/chan_sip.c: Merged revisions 53079 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 - lines Cleaning up the devicestate callback function ........ - -2007-02-01 20:14 +0000 [r53076-53078] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 53075 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53075 | tilghman | 2007-02-01 14:09:52 -0600 - (Thu, 01 Feb 2007) | 10 lines Merged revisions 53074 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 - Feb 2007) | 2 lines Bug 8965 - Allow FIELDQTY to work with both - variables and dialplan functions ........ ................ - -2007-02-01 19:34 +0000 [r53073] Joshua Colp <jcolp@digium.com> - - * /, main/asterisk.c: Merged revisions 53072 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53072 | file | 2007-02-01 13:33:33 -0600 (Thu, 01 Feb 2007) | 2 - lines Add missing 'F' letter to getopt so it magically becomes a - valid option. (issue #8960 reported by tzafrir) ........ - -2007-02-01 19:27 +0000 [r53071] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53070 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53070 | tilghman | 2007-02-01 13:21:20 -0600 - (Thu, 01 Feb 2007) | 10 lines Merged revisions 53069 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 - Feb 2007) | 2 lines No wonder FIELDQTY doesn't work with - functions... the documentation in pbx.c was wrong ........ - ................ - -2007-02-01 19:04 +0000 [r53067] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Signal HOLD status to phones that subscribe - for status. - -2007-02-01 17:42 +0000 [r53065-53066] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 53064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 - lines Fix silly logic. We really want to write UDPTL frames out - when the call is up. ........ - - * main/db1-ast/hash/hash.c: Make trunk compile under dev mode. - -2007-02-01 16:42 +0000 [r53063] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Merged revisions 53062 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb - 2007) | 2 lines Add explanation of port= in combination with - defaultip= (thanks jsmith) ........ - -2007-02-01 14:43 +0000 [r53061] Russell Bryant <russell@digium.com> - - * apps/app_rpt.c: Remove duplicate calls to pthread_attr_destroy() - that I put in yesterday by accident. - -2007-02-01 11:16 +0000 [r53058-53059] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/chan_h323.c: Oops -- Merged revisions 53057 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) | - 1 line chan_h323 is very stable, so let it built by default - ........ - -2007-02-01 00:38 +0000 [r53054] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Formatting changes - -2007-02-01 00:24 +0000 [r53051-53053] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 53052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 - lines When going on hold have the side that was put on hold - reinvite back to Asterisk. When going off hold have the side that - was taken off hold reinvited back to the other party. ........ - - * /, main/rtp.c: Merged revisions 53050 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 - lines Add more frame types to forward in the RTP bridge loops. - ........ - -2007-01-31 21:35 +0000 [r52905-53047] Russell Bryant <russell@digium.com> - - * main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, - channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c, - main/cdr.c, main/manager.c, pbx/pbx_spool.c, - channels/chan_skinny.c, channels/chan_h323.c, main/http.c, - pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c: Merged - revisions 53046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53046 | russell | 2007-01-31 15:32:08 -0600 - (Wed, 31 Jan 2007) | 11 lines Merged revisions 53045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 - Jan 2007) | 3 lines Fix a bunch of places where - pthread_attr_init() was called, but pthread_attr_destroy() was - not. ........ ................ - - * /, apps/app_userevent.c: Merged revisions 53042 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53042 | russell | 2007-01-31 12:18:25 -0600 (Wed, 31 Jan 2007) | - 2 lines Remove an extra \r\n from manager user events. (issue - #8955, mnicholson) ........ - - * /, main/rtp.c: Merged revisions 53040 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r53040 | russell | 2007-01-31 11:45:05 -0600 - (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 - Jan 2007) | 3 lines Use the proper format string to print - unsigned values in the rtp debug output. (issue #8954, wmis) - ........ ................ - - * /, apps/app_queue.c: Merged revisions 53037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53037 | russell | 2007-01-31 11:39:28 -0600 (Wed, 31 Jan 2007) | - 3 lines Only changed the paused status in an existing queue - member if the paused column exists. ........ - - * /, apps/app_queue.c: Merged revisions 53035 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53035 | russell | 2007-01-31 11:34:22 -0600 (Wed, 31 Jan 2007) | - 4 lines Instead of always creating a realtime queue member as - unpaused, read the "paused" column and use that value for the - paused status of the member. (issue #8949, jmls) ........ - - * /, contrib/init.d/rc.suse.asterisk: Merged revisions 53001 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r53001 | russell | 2007-01-30 17:38:42 -0600 (Tue, 30 Jan 2007) | - 2 lines Update init script for SuSE 10. (issue #8363, johnlange) - ........ - - * /, doc/cdrdriver.txt: Merged revisions 52999 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52999 | russell | 2007-01-30 17:30:34 -0600 (Tue, 30 Jan 2007) | - 2 lines Add documentation for using cdr_pgsql. (issue #8942, - lters) ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - codecs/codec_gsm.c: Merged revisions 52997 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) | - 5 lines When we are checking for a system installed version of - libgsm, we need to check for gsm.h as well. Furthermore, when - checking for this header, it may be located in a gsm/ sub - directory, so check for that, as well. (issue #8773) ........ - - * /, channels/chan_sip.c: Merged revisions 52952 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | - 5 lines Only set the DTMF flag on the rtp structure if the DTMF - mode is actually RFC2833, not just that it is not INFO. This - makes it get set for inband DTMF as well, which is not valid. - (issue #8936) ........ - - * /, main/asterisk.c: Merged revisions 52904 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52904 | russell | 2007-01-30 11:19:39 -0600 - (Tue, 30 Jan 2007) | 17 lines Merged revisions 52903 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 - Jan 2007) | 9 lines The SIGHUP handler was implemented to allow - admins to send SIGHUP to a running Asterisk process to reload the - configuration. However, doing the actual reload in the signal - handler itself is a very bad thing to do, because the reload - process includes calling non-reentrant functions such as - malloc/calloc/etc. If Asterisk is running in the background, then - the reload will happen immediately. However, if running in - console mode, the reload doesn't work until something is typed at - the console. That sort of defeats the purpose, but I don't see an - easy way to get around it at this point. ........ - ................ - -2007-01-30 15:39 +0000 [r52858-52860] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Use provided variable for name instead of - one in the structure since the structure was just allocated and - will be NULL. (issue #8938 reported by st41ker) - -2007-01-30 09:13 +0000 [r52818-52820] Paul Cadach <paul@odt.east.telecom.kz> - - * /, res/res_odbc.c: Merged revisions 52808 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52808 | pcadach | 2007-01-30 00:34:26 -0800 (Втр, 30 Янв 2007) | - 1 line Don't play with free()'d pointers ........ - - * /, configure, acinclude.m4: Merged revisions 52807 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r52807 | pcadach | 2007-01-30 00:33:22 -0800 (Втр, 30 - Янв 2007) | 1 line Handle non-standard OpenH323/PWLib library - names ........ - -2007-01-30 00:16 +0000 [r52764] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 52763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52763 | russell | 2007-01-29 18:15:50 -0600 - (Mon, 29 Jan 2007) | 13 lines Merged revisions 52762 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 - Jan 2007) | 5 lines Fix the extraction of the timestamp from - video frames. It was using the mapping for a mini-frame instead - of a video-frame, which caused it to get invalid data. (issue - #8795, mihai) ........ ................ - -2007-01-29 23:45 +0000 [r52718] Joshua Colp <jcolp@digium.com> - - * /, apps/app_mixmonitor.c: Merged revisions 52717 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, - 29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 - lines Now that filename is part of the structure and since it - comes before postprocess... we have to add it to our postprocess - line. (reported on asterisk-dev by Boris Bakchiev) ........ - ................ - -2007-01-29 22:58 +0000 [r52692-52696] Russell Bryant <russell@digium.com> - - * /, main/Makefile: Merged revisions 52695 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) | - 2 lines Add a missing quotation mark. This was pointed out by - jcmoore on #asterisk-dev. ........ - - * main/manager.c, /: Merged revisions 52688 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52688 | russell | 2007-01-29 16:55:41 -0600 (Mon, 29 Jan 2007) | - 3 lines Remove a recursive lock of the manager session. This was - pointed out by zandbelt in issue #8711. ........ - -2007-01-29 22:13 +0000 [r52680] Tilghman Lesher <tlesher@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 52679 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52679 | tilghman | 2007-01-29 16:12:12 -0600 (Mon, 29 Jan 2007) - | 2 lines Argument number correction ........ - -2007-01-29 21:37 +0000 [r52646-52648] Russell Bryant <russell@digium.com> - - * /, main/Makefile: Merged revisions 52647 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) | - 3 lines ASTLDFLAGS needs to be passed to the editline configure - script as LDFLAGS. (issue #8928, zandbelt) ........ - - * /, main/rtp.c: Merged revisions 52645 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | - 6 lines Fix a problem with packet-to-packet bridging and DTMF - mode translation. P2P bridging can only be used when the DTMF - modes don't match if the core is monitoring DTMF in both - directions. Then, the core will handle the translation. - Otherwise, this bridging method can not be used. (issue #8936) - ........ - -2007-01-29 21:03 +0000 [r52635] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Only use locking for bridge information if intense - P2P bridging is enabled. - -2007-01-29 20:51 +0000 [r52612-52613] Russell Bryant <russell@digium.com> - - * main/manager.c, /: The changes for trunk are less extensive, but - include - changing the actionlock to a rwlock - not locking the - session before doing the action callback The crash issue in 8711 - should not be an issue here. Merged revisions 52611 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r52611 | russell | 2007-01-29 14:39:20 -0600 (Mon, 29 - Jan 2007) | 10 lines The session lock can not be held while - calling action callbacks. If so, then when the WaitEvent callback - gets called, then no event can happen because the session can't - be locked by another thread. Also, the session needs to be locked - in the HTTP callback when it reads out the output string. This - fixes the deadlock reported in both 8711 and 8934. Regarding - issue 8711, there still may be an issue. If there is a second - action requested before the processing of the first action is - finished, there could still be some corruption of the output - string buffer used to build the result. (issue #8711, #8934) - ........ - - * apps/app_voicemail.c: Resolve some warnings when not building - with IMAP_STORAGE - -2007-01-29 20:22 +0000 [r52580-52610] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Change vmstates list to use linked list - macros. - - * apps/app_voicemail.c: Code cleanup of IMAP storage support in - app_voicemail. - - * /, apps/app_voicemail.c: Merged revisions 52572 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52572 | file | 2007-01-29 13:59:41 -0500 (Mon, 29 Jan 2007) | 2 - lines Use ast_calloc instead of malloc. ........ - -2007-01-29 17:49 +0000 [r52524-52525] Joshua Colp <jcolp@digium.com> - - * CHANGES, main/cli.c: Add core show channels count CLI command. - (issue #8932 reported by mr_mehul_shah) - - * /, apps/app_voicemail.c: Merged revisions 52523 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52523 | file | 2007-01-29 12:33:19 -0500 (Mon, 29 Jan 2007) | 2 - lines Set quota information to 0 when creating a vm_state. (issue - #8924 reported by neutrino88) ........ - -2007-01-29 17:03 +0000 [r52522] Russell Bryant <russell@digium.com> - - * /, main/jitterbuf.c, include/jitterbuf.h: Merged revisions - 52494,52506 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) | - 4 lines Fixed problem with jitterbuf, whereas it would not - complain about, and would allow itself to be overfilled (per the - max_jitterbuf parameter). Now it rejects any data over and above - that size, and complains about it. ........ r52506 | russell | - 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines Clean up a - few things in the last commit to the adaptive jitterbuffer code. - - Specifically indicate to the compiler that the "dropem" - variable only needs one but. - Change formatting to conform to - coding guidelines. ........ - -2007-01-28 05:18 +0000 [r52463] Tilghman Lesher <tlesher@digium.com> - - * /, configure, configure.ac: Merged revisions 52462 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r52462 | tilghman | 2007-01-27 23:15:07 -0600 (Sat, 27 - Jan 2007) | 2 lines Suggested change to fix normal usage of - --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing - list) ........ - -2007-01-27 02:15 +0000 [r52332-52417] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 52416 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52416 | file | 2007-01-26 21:13:41 -0500 (Fri, - 26 Jan 2007) | 10 lines Merged revisions 52415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 - lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log - follow documentation. (issue #7677 reported by amilcar) ........ - ................ - - * /, channels/chan_iax2.c: Merged revisions 52370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52370 | file | 2007-01-26 19:08:18 -0500 (Fri, - 26 Jan 2007) | 10 lines Merged revisions 52360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 - lines Make the last context entry read in the dominant one. - (issue #8918 reported by pj) ........ ................ - - * /, main/file.c: Merged revisions 52335 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52335 | file | 2007-01-26 18:46:47 -0500 (Fri, 26 Jan 2007) | 2 - lines Fix core show file formats CLI command. ........ - - * main/file.c, main/image.c: Convert some more stuff to read/write - lists. - -2007-01-25 22:49 +0000 [r52168-52308] Joshua Colp <jcolp@digium.com> - - * CHANGES, main/db.c: Add DBDel and DBDelTree manager commands. - (issue #8516 reported by dprado) - - * /, main/jitterbuf.c: Merged revisions 52265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52265 | file | 2007-01-25 14:18:33 -0500 (Thu, - 25 Jan 2007) | 10 lines Merged revisions 52264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 - lines Allow dequeueing of frames with negative timestamp by - moving jitterbuffer frames check to jb_next. (issue #8546 - reported by harmen) ........ ................ - - * channels/chan_sip.c: Use atomic operation functions for - use/ringing/hold manipulation. - - * /, channels/chan_sip.c: Merged revisions 52210 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 - lines Drop out variables I accidentally put in. ........ - - * /, channels/chan_sip.c: Merged revisions 52208 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 - lines Decrement onHold count if we are hung up on and still on - hold. (issue #8909 reported by alexh42) ........ - - * /, apps/app_mixmonitor.c: Merged revisions 52163 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, - 24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 - lines Add another note about audio files being played back to - each bridged party. (issue #8718 reported by ppyy) ........ - ................ - -2007-01-25 01:38 +0000 [r52108-52161] Russell Bryant <russell@digium.com> - - * configs/users.conf.sample, /, apps/app_voicemail.c: Merged - revisions 52160 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52160 | russell | 2007-01-24 19:37:16 -0600 (Wed, 24 Jan 2007) | - 2 lines By suggestion from kpfleming last week, change - "vmpassword" to "vmsecret". ........ - - * /, include/asterisk/dial.h: Merged revisions 52107 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24 - Jan 2007) | 3 lines Fix the formatting of doxygen comments to - properly indicate that the comment documents the previous entity, - as opposed to the next one. ........ - -2007-01-24 20:35 +0000 [r52053-52086] Steve Murphy <murf@digium.com> - - * UPGRADE.txt, apps/app_chanisavail.c: As per bug 8859 (Add option - to revert old ChanIsAvail() with 's' option behavior), this - update makes the 't' option available, which calls - ast_parse_device_state instead of ast_device_state. This option - will not dive into the channel driver to find the status of the - device (which could be good if sip devicestate isn't returning - full status, for various reasons). - - * utils/Makefile, /, utils/check_expr.c: Merged revisions 52052 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r52052 | murf | 2007-01-24 11:26:22 -0700 (Wed, - 24 Jan 2007) | 9 lines Merged revisions 52002 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 - line updated check_expr via 8322 (refactoring of expression - checking impl); elfring contributed a nice code reorg, I - contributed some time to get it working again, better messages - ........ ................ - -2007-01-24 18:23 +0000 [r52025-52050] Joshua Colp <jcolp@digium.com> - - * main/dial.c (added), /, apps/app_page.c, main/Makefile, - include/asterisk/dial.h (added): Merged revisions 52049 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 - lines Merge in dialing API and the app_page that uses it. (issue - #BE-118) ........ - - * /, channels/chan_sip.c: Merged revisions 52016 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 - lines Fix changing channel formats when joint capability changes - and there are no audio formats... I didn't break it originally! - (issue #8535 reported by ivoc) ........ - -2007-01-24 09:42 +0000 [r51905-51933] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 51931 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 - lines Show capabilities *and* preference in general settings in - "sip show settings" (reported by Clona/Telio - Thanks!) ........ - - * include/asterisk/http.h, main/http.c: Doxygen updates - - * funcs/func_rand.c, funcs/func_base64.c, funcs/func_module.c, - funcs/func_md5.c, funcs/func_db.c, funcs/func_version.c, - funcs/func_timeout.c, funcs/func_env.c, funcs/func_math.c, - funcs/func_strings.c, funcs/func_sha1.c, funcs/func_logic.c, - funcs/func_uri.c, funcs/func_global.c, funcs/func_enum.c, - funcs/func_groupcount.c, funcs/func_odbc.c, funcs/func_shell.c, - funcs/func_channel.c, funcs/func_cdr.c, funcs/func_callerid.c: - Doxygen update - - * main/udptl.c: Adding some doxygen for udptl.c - -2007-01-24 01:00 +0000 [r51850] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 51848 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51848 | russell | 2007-01-23 18:59:58 -0600 - (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 - Jan 2007) | 6 lines Fix an issue related to synchronization of - recordings when using Monitor(). The bug is a miscalculation of - the amount to seek the stream for writing to disk when the number - of samples coming in and out of a channel do not match up. (issue - #8298, #8887, report and patch by guillecabeza, patch files - created and testing done by whoiswes) ........ ................ - -2007-01-24 00:22 +0000 [r51831] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Close file after we do the translation, and map - memory for both reading/writing. (issue #8886 reported by - cwegener) - -2007-01-24 00:21 +0000 [r51830] Russell Bryant <russell@digium.com> - - * /, apps/app_while.c: Merged revisions 51829 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51829 | russell | 2007-01-23 18:19:55 -0600 - (Tue, 23 Jan 2007) | 12 lines Merged revisions 51828 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 - Jan 2007) | 4 lines Don't set a new value for the END_ variable - on the channel before using the old value. If you do, it will - lead to accessing a memory address that has been free()'d. (issue - #8895, arkadia) ........ ................ - -2007-01-23 22:59 +0000 [r51801] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, channels/chan_zap.c, /, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_alsa.c, - channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c: - Merged revisions 51788 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 - lines Update channel drivers to use module referencing so that - unloading them while in use will not result in crashes. (issue - #8897 reported by junky) ........ - -2007-01-23 22:09 +0000 [r51751-51787] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 51781 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) | - 6 lines Fix some bugs in process_message(). The manager session - lock needs to be held when sending some sort of response, or - calling one of the manager action callbacks. This resolves an - issue where people using the GUI would get random crashes when - they start clicking around a lot. (issue #8711, reported and - debugged by zandbelt) ........ - - * main/manager.c, /: Merged revisions 51750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) | - 4 lines When traversing the list of manager actions, the iterator - needs to be initialized to the list head *after* locking the - list. Also, lock the actions list in one place it is being - accessed where it was not being done. ........ - -2007-01-23 20:36 +0000 [r51684-51717] Steve Murphy <murf@digium.com> - - * /, res/res_features.c: Merged revisions 51716 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51716 | murf | 2007-01-23 13:32:54 -0700 (Tue, 23 Jan 2007) | 1 - line this mod from 8593 (dstchannel in cdr is empty when transfer - call). ........ - - * /, main/callerid.c: Merged revisions 51683 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1 - line via 8748 (callerid.c loses name when returning - PRIVATE_NUMBER flag), the user suggested this mod, saying it - would allow 'WITHHELD' to appear in the name field, which would - be useful ........ - -2007-01-23 15:36 +0000 [r51659] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8817 - Registry corruption when - packet retransmits fail. (tootai, patchy by oej) - -2007-01-23 06:56 +0000 [r51623] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/chan_h323.c, channels/Makefile: Merged revisions - 51615 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | - 1 line Do not abort Asterisk startup if h323 configuration file - not found (reported by mithraen) ........ - -2007-01-23 04:45 +0000 [r51463-51592] Joshua Colp <jcolp@digium.com> - - * doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Make 'H' - command do as advertised and add 'E' and 'V' commands to - ExternalIVR. (issue #8165 reported by mnicholson) - - * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add SRV - Lookup support on outbound calls to chan_iax2. It's listed in the - RFC so we might want to support it and please don't hurt me Marko - ... (issue #7812 reported by drorlb) - - * /, channels/chan_sip.c: Merged revisions 51558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 - lines Only change audio formats on the channel if we have an - audio format to change to. (issue #8535 reported by ivoc) - ........ - - * /: No more conflicts on properties! svnmerge-block be gone! - - * /, res/res_musiconhold.c: Merged revisions 51513 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51513 | file | 2007-01-22 20:45:04 -0500 (Mon, - 22 Jan 2007) | 10 lines Merged revisions 51512 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan 2007) | 2 - lines Yield before reading from zaptel timing source under - Solaris so that other threads get a chance to do things. (issue - #7875 reported by bob) ........ ................ - - * main/autoservice.c: Might as well go crazy here too and make the - autoservice list read/write. - - * main/pbx.c, main/autoservice.c, main/frame.c, main/say.c, - main/jitterbuf.c, main/devicestate.c, main/utils.c, main/enum.c, - main/fskmodem.c, main/config.c, main/cli.c, main/io.c, - main/channel.c, main/cdr.c, main/abstract_jb.c, main/logger.c, - main/callerid.c, main/file.c, main/app.c, main/image.c, - main/alaw.c, main/asterisk.c, main/dsp.c: Cosmetic changes. Make - main source files better conform to coding guidelines and - standards. (issue #8679 reported by johann8384) - - * main/rtp.c: Change RTP protos list to be read/write. Most of the - time it's only going to be read so making it use mutex locks was - a waste. - - * main/rtp.c: Make the RTP stack better conform to coding - guidelines. (issue #8679 reported by johann8384) - -2007-01-22 19:42 +0000 [r51413] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 51409 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51409 | murf | 2007-01-22 12:28:51 -0700 (Mon, 22 Jan 2007) | 1 - line This fixes 8836, according to dnatural ........ - -2007-01-22 19:22 +0000 [r51408] Joshua Colp <jcolp@digium.com> - - * /, apps/app_mixmonitor.c: Merged revisions 51407 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, - 22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 - lines Move filestream creation to Mixmonitor loop. This will - prevent a blank file from being created if no frames ever pass - through to be recorded. (issue #7589 reported by steve_mcneil) - ........ ................ - -2007-01-22 19:00 +0000 [r51405] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove (to quote Rizzo) "useless" variable. - -2007-01-21 03:25 +0000 [r51353] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Fix bug introduced during constification (reported by - tzanger via IRC) - -2007-01-20 18:27 +0000 [r51352] Russell Bryant <russell@digium.com> - - * include/asterisk/frame.h: Add a comment that the frame type - constants are transmitted directly over IAX2. - -2007-01-20 06:54 +0000 [r51349-51351] Jason Parker <jparker@digium.com> - - * /, configs/say.conf.sample: Merged revisions 51350 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan - 2007) | 5 lines Fix Italian numeral support in say.conf for - "_[2-9]00" case. "2131" would've translated to something along - the lines of (pardon my..Italian {or lack thereof}) - "duecentocentotrentuno", which makes no sense at all. ........ - - * /, configs/say.conf.sample: Merged revisions 51348 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan - 2007) | 8 lines Fix German language support in say.conf Properly - support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the - same format as zweiundzwanzig (as do all other "_ZX" spoken - numerals) Fix support for numbers in the 10,000,000 to 99,999,999 - range. Add support for numbers in the 100,000,000 to 999,999,999 - range. ........ - -2007-01-20 00:13 +0000 [r51314-51344] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 51343 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51343 | russell | 2007-01-19 18:13:06 -0600 (Fri, 19 Jan 2007) | - 2 lines Remove an unused instance of an unnamed enum. ........ - - * /, apps/app_meetme.c: Merged revisions 51341 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51341 | russell | 2007-01-19 16:19:10 -0600 (Fri, 19 Jan 2007) | - 2 lines Remove another duplicated definition ........ - - * /, apps/app_meetme.c: Merged revisions 51339 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51339 | russell | 2007-01-19 15:20:20 -0600 (Fri, 19 Jan 2007) | - 2 lines Remove a variable that was declared twice. ........ - - * /, codecs/gsm/Makefile: Merged revisions 51331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51331 | russell | 2007-01-19 13:30:54 -0600 (Fri, 19 Jan 2007) | - 3 lines Add a couple more processors that need optimizations - excluded. (issue #8637) ........ - - * /, channels/chan_gtalk.c: Merged revisions 51328 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 - Jan 2007) | 5 lines Fix VLDTMF support in chan_gtalk. - AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same - thing. So, a digit would have been interpreted incorrectly here. - Since the channel driver will always have the begin and end - callbacks called for a digit, only support the button-down and - button-up messages. ........ - - * /, .cleancount: Merged revisions 51326 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51326 | russell | 2007-01-19 13:02:55 -0600 (Fri, 19 Jan 2007) | - 2 lines Bump the cleancount since my last commit changed the - channel structure. ........ - - * channels/chan_zap.c, channels/chan_local.c, main/frame.c, /, - channels/chan_sip.c, channels/chan_agent.c, - include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_iax2.c, channels/chan_oss.c, main/rtp.c, - main/channel.c, channels/chan_jingle.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_mgcp.c: Merged revisions - 51311 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | - 23 lines Merge the changes from the /team/group/vldtmf_fixup - branch. The main bug being addressed here is a problem introduced - when two SIP channels using SIP INFO dtmf have their media - directly bridged. So, when a DTMF END frame comes into Asterisk - from an incoming INFO message, Asterisk would try to emulate a - digit of some length by first sending a DTMF BEGIN frame and - sending a DTMF END later timed off of incoming audio. However, - since there was no audio coming in, the DTMF_END was never - generated. This caused DTMF based features to no longer work. To - fix this, the core now knows when a channel doesn't care about - DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If - this is the case, then Asterisk will not emulate a digit of some - length, and will instead just pass through the single DTMF END - event. Channel drivers also now get passed the length of the - digit to their digit_end callback. This improves SIP INFO support - even further by enabling us to put the real digit duration in the - INFO message instead of a hard coded 250ms. Also, for an incoming - INFO message, the duration is read from the frame and passed into - the core instead of just getting ignored. (issue #8597, maybe - others...) ........ - -2007-01-19 18:00 +0000 [r51308-51312] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/strings.h: As the comment in the diff says: - AST_INLINE_API() is a macro that takes a block of code as an - argument. Using preprocessor #directives in the argument is not - supported by all compilers, and it is a bit of an obfuscation - anyways, so avoid it. As a workaround, define a macro that - produces either its argument or nothing, and use that instead of - #ifdef/#endif within the argument to AST_INLINE_API(). - - * main/rtp.c: in the interest of portability, avoid using %zd when - all we need is to print is an integer that fits in 16 bits. - - * channels/chan_iax2.c: sizeof() is compatible with format %d so - don't be too picky on printf formats. - - * channels/chan_zap.c: remove variable declaration in the middle of - a block - -2007-01-19 17:19 +0000 [r51303-51305] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in: Regenerate configure - script to reflect recent zaptel changes - - * include/asterisk/zapata.h: Include tonezone.h for linux, too - - * main/asterisk.c: Merged revisions 51302 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51302 | russell | 2007-01-19 10:56:17 -0600 - (Fri, 19 Jan 2007) | 12 lines Merged revisions 51300 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 - Jan 2007) | 4 lines Fix a memory leak on command line tab - completion. The container for the matches was freed, but the - individual matches themselves were not. (issue #8851, arkadia) - ........ ................ - -2007-01-19 16:51 +0000 [r51297-51301] Luigi Rizzo <rizzo@icir.org> - - * main/Makefile: forgot to add AST_LIBS += $(BKTR_LIB) - - * main/channel.c: include "asterisk/zapata.h" to get the zaptel - headers. this should be the last one left around... - - * channels/chan_zap.c: whoops, fix a cut&paste error... - - * channels/chan_zap.c: slight change to the initialization of a - structure, also using '\0' to make it clear we need a (char)0 - -2007-01-19 16:30 +0000 [r51296] Russell Bryant <russell@digium.com> - - * main/manager.c: Break out of the config processing loop for - manager.conf once the correct user has been found so that 'cat' - is non-NULL. This way, users.conf is only checked when necessary. - (issue #8852, akohlsmith, committed patch a bit different) - -2007-01-19 16:28 +0000 [r51285-51295] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_zap.c: include "asterisk/zapata.h" to get the - zaptel headers. - - * codecs/codec_zap.c: include "asterisk/zapata.h" to get the zaptel - headers - - * apps/app_meetme.c: include "asterisk/zapata.h" instead of testing - for the location of the header files. On passing, add a cast to - insure -Werror clean compilation on FreeBSD 6.x, where time_t - does not match %ld - - * apps/app_zapbarge.c, apps/app_flash.c, apps/app_zapscan.c, - apps/app_zapras.c, res/res_musiconhold.c, channels/chan_iax2.c, - apps/app_rpt.c: include "asterisk/zapata.h" instead of looking - directly for the zaptel.h and tonezone.h - - * configure.ac: another freebsd-specific check for zaptel - compatibility - - * include/asterisk/zapata.h (added): Add a stub file to find the - zaptel headers in the right place, rather than repeating the - check on every single file. Changes to the individual files are - coming. The header file name has been suggested by kevin. - Approved by: kpfleming - - * makeopts.in: forgot to add BKTR_INCLUDE and BKTR_LIB in - makeopts.in - - * configure.ac: add comments that AC_USE_SYSTEM_EXTENSIONS and - AST_PROG_LD do not work on FreeBSD - presumably they depend on - some auto* feature that is not installed by default. I am not - sure on what is a proper fix. In my local copy i simply comment - them out. The AST_PROG_LD is a long standing isse, there were - attempts to fix it in the past but probably not enough has been - copied to acinclude.m4, and i had forgotten about it because i - commented out this call in configure.ac long ago - - * configure.ac: Add check for backtrace support on platforms that - do not have it natively. Part of it leaked in in a previous - commit. - - * configure.ac: remove a useless (and harmful on some platforms) - -lnsl from IKSEMEL_LIB. Actually i am not even sure whether - -lgcrypt -lgpg-error are needed. - - * configure.ac: simplify checking for zaptel version and location - (for linux, this is functionally equivalent to the previous - method; for FreeBSD, it re-adds inspection in $PREFIX/zaptel.h). - Please wait to regenerate the "configure" file as i have another - few pending changes to configure.ac Not applicable to 1.4 until - acinclude.m4 is also updated. - -2007-01-19 00:28 +0000 [r51273-51275] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c, /: Merged revisions 51274 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) - | 3 lines chan_zap compiles without libpri after committing 7877 - patch ........ - - * channels/chan_zap.c, /: Merged revisions 51272 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51272 | dhubbard | 2007-01-18 17:56:49 -0600 - (Thu, 18 Jan 2007) | 11 lines Merged revisions 51271 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 - Jan 2007) | 3 lines issue 7877: chan_zap module reload does not - use default/initialized values on subsequent loads. Reset - configuration variables to default values prior to parsing - configuration file. ........ ................ - -2007-01-18 22:56 +0000 [r51266] Jason Parker <jparker@digium.com> - - * main/pbx.c, /, funcs/func_strings.c, apps/app_voicemail.c: Merged - revisions 51265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51265 | qwell | 2007-01-18 16:50:23 -0600 (Thu, 18 Jan 2007) | 4 - lines Add some more checks for option_debug before - ast_log(LOG_DEBUG, ...) calls. Issue 8832, patch(es) by tgrman - ........ - -2007-01-18 21:57 +0000 [r51263] Russell Bryant <russell@digium.com> - - * Makefile, /, configure, main/Makefile, acinclude.m4, makeopts.in: - Merged revisions 51262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | - 5 lines Ensure that the locations given to the Asterisk configure - script for ncurses, curses, termcap, or tinfo are further passed - along to the editline configure script. This fixes some - cross-compilation environments. (issue #8637, reported by ovi, - patch by me) ........ - -2007-01-18 21:15 +0000 [r51257] Tilghman Lesher <tlesher@digium.com> - - * /, main/stdtime/localtime.c: Merged revisions 51256 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51256 | tilghman | 2007-01-18 15:14:24 -0600 - (Thu, 18 Jan 2007) | 10 lines Merged revisions 51255 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 - Jan 2007) | 2 lines If a timezone is not specified, assume - localtime (instead of gmtime) (Issue #7748) ........ - ................ - -2007-01-18 19:19 +0000 [r51252] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 51251 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan - 2007) | 2 lines Only start timeout once we reach the end of the - files to play back. ........ - -2007-01-18 19:03 +0000 [r51249] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix filename completion for "module load" and "load" - CLI commands. Issue 8846 - -2007-01-18 18:54 +0000 [r51247] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix trunk version of manager support for - users.conf. Now it actually pays attention to the "hasmanager" - option. (Thanks to Anthony L. for pointing out that this was - broken!) - -2007-01-18 18:39 +0000 [r51244] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 51243 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 - lines Copy MOH settings when calling a peer so that if they put - someone on hold or get put on hold themselves they get the right - music class. (issue #8840 reported by mdu113) ........ - -2007-01-18 18:36 +0000 [r51242] Jason Parker <jparker@digium.com> - - * main/channel.c, /: Merged revisions 51241 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2 - lines Fix an issue with deprecated commands ........ - -2007-01-18 17:52 +0000 [r51237] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/vmdb.sql, /: Merged revisions 51236 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51236 | tilghman | 2007-01-18 11:49:41 -0600 - (Thu, 18 Jan 2007) | 10 lines Merged revisions 51235 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 - Jan 2007) | 2 lines Document all the fields, including the - indication that "uniqueid" should not be renamed. ........ - ................ - -2007-01-18 17:33 +0000 [r51234] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 51233 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51233 | russell | 2007-01-18 11:18:43 -0600 (Thu, 18 Jan 2007) | - 3 lines Make the "hasmanager" option in users.conf actually have - an effect. (issue #8740, LnxPrgr3) ........ - -2007-01-18 06:59 +0000 [r51221] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Update ast_append_ha() usage - -2007-01-18 05:24 +0000 [r51212-51215] Joshua Colp <jcolp@digium.com> - - * apps/app_page.c, CHANGES: Add 's' option to Page application - which checks devicestate before dialing. (issue #8673 reported by - sunder) - - * /, apps/app_voicemail.c: Merged revisions 51213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51213 | file | 2007-01-17 19:48:55 -0500 (Wed, 17 Jan 2007) | 2 - lines Build the IMAP remote directory string better and properly. - Fix an issue with encoding the GSM voicemail when attaching to - the voicemail. (issue #8808 reported by akohlsmith) ........ - - * /, main/rtp.c: Merged revisions 51211 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 - lines Pass data as well for hold/unhold/vidupdate frames. (issue - #8840 reported by mdu113) ........ - -2007-01-17 23:35 +0000 [r51199-51207] Russell Bryant <russell@digium.com> - - * /, funcs/func_odbc.c: Merged revisions 51205 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51205 | russell | 2007-01-17 17:31:11 -0600 (Wed, 17 Jan 2007) | - 5 lines Fix some instances where when loading func_odbc, a - double-free could occur. Also, remove an unneeded error message. - If the failure condition is actually a memory allocation failure, - a log message will already be generated automatically. ........ - - * channels/chan_zap.c, /: Merged revisions 51204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | - 4 lines Instead of dividing the offset by 2 directly, make it - more clear that the offset is being scaled by the size of the - elements in the buffer. (Inspired by a discussing on the - asterisk-dev list about this code) ........ - - * /, channels/chan_sip.c: Merged revisions 51198 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51198 | russell | 2007-01-17 15:18:35 -0600 - (Wed, 17 Jan 2007) | 11 lines Merged revisions 51197 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 - Jan 2007) | 3 lines Move the check for a failure of - ast_channel_alloc() to before locking the pvt structure again. - Otherwise, on a failure, this will cause a deadlock. ........ - ................ - -2007-01-17 20:57 +0000 [r51196] Tilghman Lesher <tlesher@digium.com> - - * /, main/utils.c: Merged revisions 51195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51195 | tilghman | 2007-01-17 14:56:15 -0600 - (Wed, 17 Jan 2007) | 12 lines Merged revisions 51194 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 - Jan 2007) | 4 lines When ast_strip_quoted was called with a - zero-length string, it would treat a NULL as if it were the - quoting character (and would thus return the string in memory - immediately following the passed-in string). ........ - ................ - -2007-01-17 19:43 +0000 [r51193] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Don't hold channel lock while sleeping/waiting - for audio stream to get setup. (issue #8834 reported by phsultan) - -2007-01-17 17:37 +0000 [r51189] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 51186 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51186 | qwell | 2007-01-17 11:36:53 -0600 (Wed, 17 Jan 2007) | 2 - lines re-add "password" for realtime voicemail ........ - -2007-01-17 06:37 +0000 [r51183] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 51182 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 - lines Return the correct result when directly writing out a - packet so that the core doesn't then decide to handle it the - regular way again. (issue #8833 reported by rcourtna) ........ - -2007-01-17 01:30 +0000 [r51177] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 51176 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51176 | kpfleming | 2007-01-16 19:29:12 -0600 (Tue, 16 Jan 2007) - | 2 lines a few more coding style cleanups and one bug fix (from - AnthonyL) ........ - -2007-01-17 00:50 +0000 [r51173] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 51172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2 - lines Move rescheduling of lagrq/pings into the scheduler - callback. ........ - -2007-01-17 00:22 +0000 [r51166-51171] Jason Parker <jparker@digium.com> - - * /, main/rtp.c: Merged revisions 51170 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4 - lines Fix issue with dtmf continuation packets when the dtmf - digit is 0... Issue 8831 ........ - - * contrib/scripts/vmdb.sql, /, apps/app_voicemail.c: Merged - revisions 51167 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51167 | qwell | 2007-01-16 16:50:19 -0600 (Tue, 16 Jan 2007) | 6 - lines Fix an issue with IMAP storage and realtime voicemail. Also - update the vmdb sql script for IMAP specific options. Issue 8819, - initial patches by bsmithurst (slightly modified by me) ........ - - * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 51165 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51165 | qwell | 2007-01-16 16:07:53 -0600 (Tue, 16 Jan 2007) | 2 - lines change documentation to reflect new procedure in 1.4/trunk - ........ - -2007-01-16 21:52 +0000 [r51160-51163] Tilghman Lesher <tlesher@digium.com> - - * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions - 51162 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51162 | tilghman | 2007-01-16 15:51:15 -0600 - (Tue, 16 Jan 2007) | 10 lines Merged revisions 51161 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 - Jan 2007) | 2 lines Add documentation walkthrough on getting - Postgres to work with voicemail (from Issue 8513) ........ - ................ - - * /, apps/app_voicemail.c: Merged revisions 51159 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51159 | tilghman | 2007-01-16 15:28:39 -0600 - (Tue, 16 Jan 2007) | 10 lines Merged revisions 51158 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 - Jan 2007) | 2 lines Postgres driver doesn't like a NULL pointer - when retrieving the length (Bug 8513) ........ ................ - -2007-01-16 19:01 +0000 [r51155] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: remove pointless DEBUG message (watch those - patch merges, people!) - -2007-01-16 17:50 +0000 [r51152] Joshua Colp <jcolp@digium.com> - - * res/res_features.c, CHANGES, configs/features.conf.sample: Add - parkedcalltransfers option for res_features. This basically - enables/disables DTMF based transfers. If you want to get former - behavior you will have to make sure it is enabled. - -2007-01-16 17:47 +0000 [r51151] Matt O'Gorman <mogorman@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 51150 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ........ r51150 - | mogorman | 2007-01-16 11:46:12 -0600 (Tue, 16 Jan 2007) | 2 - lines minor things i missed before i get jumped on ........ - -2007-01-16 17:42 +0000 [r51149] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 51148 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51148 | file | 2007-01-16 12:39:50 -0500 (Tue, - 16 Jan 2007) | 10 lines Merged revisions 51145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 - lines Return previous behavior. ParkedCalls will be able to do - DTMF based transfers again. trunk however will get an option to - allow this to be set on/off. (issue #8804 reported by nortex) - ........ ................ - -2007-01-16 17:39 +0000 [r51147] Jason Parker <jparker@digium.com> - - * /, main/file.c: Merged revisions 51146 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51146 | qwell | 2007-01-16 11:36:53 -0600 (Tue, 16 Jan 2007) | 6 - lines Display more useful output when streaming files. Include - the channel name to which the file is being played. Issue 8828, - patch by junky. ........ - -2007-01-16 17:23 +0000 [r51144] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, configs/phone.conf.sample, CHANGES: Add - support for G729 passthrough with Sigma Designs boards. (issue - #8829 reported by ywalther) - -2007-01-16 08:38 +0000 [r51123] Tilghman Lesher <tlesher@digium.com> - - * channels/iax2-parser.h, channels/iax2.h, channels/chan_iax2.c, - channels/iax2-parser.c: IAX2 remote variables - Bug 7619 - -2007-01-16 05:56 +0000 [r51090] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, /: Merged revisions 51087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, - 16 Jan 2007) | 10 lines Merged revisions 51085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 - lines Add none as a valid callgroup/pickupgroup option. I - consider it a bug that it would inherit it all the way down and - not have any way to reset it to nothing - so that's why it is in - 1.2. (issue #8296 reported by gkloepfer) ........ - ................ - -2007-01-16 01:20 +0000 [r51058-51060] Russell Bryant <russell@digium.com> - - * configs/osp.conf.sample: Fix a couple of typos in the sample - osp.conf. - - * /, main/config.c: Merged revisions 51057 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r51057 | russell | 2007-01-15 19:15:44 -0600 (Mon, 15 Jan 2007) | - 3 lines It is possible for the config pointer to be NULL here, so - it needs to be checked before dereferencing it. ........ - -2007-01-16 00:29 +0000 [r51031] Matt O'Gorman <mogorman@digium.com> - - * configs/users.conf.sample, /, apps/app_voicemail.c: Patch allows - for changing voicemail password in users.conf from voicemail - main, written by AnthonyL bug #8436 - -2007-01-15 23:51 +0000 [r50995] Russell Bryant <russell@digium.com> - - * /, Makefile.rules: Merged revisions 50994 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50994 | russell | 2007-01-15 17:49:48 -0600 (Mon, 15 Jan 2007) | - 2 lines Filter out a few CFLAGS that are not valid CXXFLAGS. - ........ - -2007-01-15 21:12 +0000 [r50958] Matt O'Gorman <mogorman@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 50957 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ................ - r50957 | mogorman | 2007-01-15 15:08:07 -0600 (Mon, 15 Jan 2007) - | 12 lines Merged revisions 50946 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 - | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 - lines Solves issue with forwarding voicemails from folders other - than inbox. patch by anthonyl. ........ ................ - -2007-01-15 18:24 +0000 [r50922] Jason Parker <jparker@digium.com> - - * /: These deprecated functions were removed in trunk on purpose. - No need to re-add. - -2007-01-15 16:40 +0000 [r50896] Joshua Colp <jcolp@digium.com> - - * main/manager.c, /: Merged revisions 50895 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50895 | file | 2007-01-15 11:36:07 -0500 (Mon, 15 Jan 2007) | 2 - lines Move event processing into do_message so that it gets - executed again when events are tripped. ........ - -2007-01-15 15:08 +0000 [r50868-50869] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, main/Makefile, - configure.ac, Makefile.rules, acinclude.m4, makeopts.in: Merged - revisions 50867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007) - | 2 lines use the ACX_PTHREAD macro from the Autoconf macro - archive for setting up compiler pthreads support... should - improve portability to platforms with unusual pthreads - requirements ........ - - * codecs/g722: ignore dependency files in this directory - -2007-01-15 02:28 +0000 [r50847] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_oss.c: Feature: allow soundcard to be used in both - modes (autoanswer and not), selectable by how it is called in the - dialplan. This allows a speaker system hooked up to the soundcard - to be used for both ring notification, as well as paging. - -2007-01-14 22:00 +0000 [r50821] Joshua Colp <jcolp@digium.com> - - * /, main/astmm.c: Merged revisions 50820 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50820 | file | 2007-01-14 16:59:05 -0500 (Sun, 14 Jan 2007) | 2 - lines Add missing newlines for two memory CLI commands. ........ - -2007-01-14 05:34 +0000 [r50783-50784] Tilghman Lesher <tlesher@digium.com> - - * main/config.c: Bug 8803 - Fix crash in API - - * /, main/db1-ast/hash/hsearch.c, main/db1-ast/btree/bt_page.c, - main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, - main/db1-ast/hash/hash.c, main/db1-ast/db/db.c, - main/db1-ast/recno/rec_get.c, main/db1-ast/btree/bt_seq.c, - main/db1-ast/hash/hash_func.c, main/db1-ast/btree/bt_utils.c, - main/db1-ast/recno/rec_seq.c, main/db1-ast/btree/bt_overflow.c, - main/db1-ast/btree/bt_search.c, main/db1-ast/btree/bt_conv.c, - main/db1-ast/btree/bt_close.c, main/db1-ast/btree/bt_put.c, - main/db1-ast/recno/rec_utils.c, main/db1-ast/hash/hash_bigkey.c, - main/db1-ast/recno/rec_open.c, main/db1-ast/recno/rec_delete.c, - main/db1-ast/hash/hash_buf.c, main/db1-ast/hash/hash_page.c, - main/db1-ast/recno/rec_close.c, main/db1-ast/recno/rec_put.c, - main/db1-ast/include/ndbm.h, main/db1-ast/btree/bt_debug.c, - main/db1-ast/mpool/mpool.c, main/db1-ast/btree/bt_split.c, - main/db1-ast/btree/bt_open.c, main/db1-ast/btree/bt_delete.c, - main/db1-ast/hash/hash_log2.c: Merged revisions 50782 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r50782 | tilghman | 2007-01-13 23:13:47 -0600 - (Sat, 13 Jan 2007) | 10 lines Merged revisions 50781 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 - Jan 2007) | 2 lines Bug 8814 - db should look for its header - using a relative path, instead of the system path (Fixes FreeWRT) - ........ ................ - -2007-01-13 16:47 +0000 [r50755] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /, build_tools/make_sample_voicemail (added): Merged - revisions 50754 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50754 | kpfleming | 2007-01-13 10:45:37 -0600 (Sat, 13 Jan 2007) - | 2 lines when building the sample greetings for maibox - 1234@default during 'make samples', build a greeting for each - language and file format the user selected to install with - menuselect (reported by Brian Capouch on asterisk-dev) ........ - -2007-01-13 06:01 +0000 [r50675-50728] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 50727 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2 - lines Only write a frame out to the channel if one exists. There - are cases where one may not and would therefore cause the channel - driver to segfault. (issue #8434 reported by slimey) ........ - - * channels/chan_sip.c: Get rid of unneeded code, fix a spelling - mistake, and use registry state a bit more. (issue #8402 reported - by rizzo) - - * configs/iax.conf.sample: Clarify what the trunkmaxsize value is - in (bytes). - - * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Drop - trunkrealloc option and just have the maximum size be a - configurable option. This is per Kevin's comments on -dev and my - own thoughts after I put the previous option in. - - * channels/chan_sip.c: Ensure error variable is set to 0 or else we - might get false error messages. (issue #8798 reported by tootai, - fix by anthonyl) - - * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Merge in - trunkrealloc option for chan_iax2. (issue #8267 reported by - marcodmb, branch by anthonyl) - - * /, res/res_snmp.c: Merged revisions 50674 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50674 | file | 2007-01-12 22:04:55 -0500 (Fri, 12 Jan 2007) | 2 - lines Only join the snmp thread on an unload if the thread is - actually running. (issue #8810 reported by junky) ........ - -2007-01-12 19:25 +0000 [r50648] Jason Parker <jparker@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 50647 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50647 | qwell | 2007-01-12 13:24:40 -0600 (Fri, 12 Jan 2007) | 2 - lines Update documentation to state that you shouldn't use - realtime static with voicemail.conf ........ - -2007-01-12 18:13 +0000 [r50603-50629] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Exit from session loop upon error (ie: they - disconnected) and don't do any buffer manipulation in do_message. - get_input will handle it. - - * main/manager.c, /: Merged revisions 50602 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50602 | file | 2007-01-12 11:42:33 -0500 (Fri, 12 Jan 2007) | 2 - lines We need to check for res being 0 in do_message itself, - otherwise our headers will get lost. ........ - -2007-01-12 15:01 +0000 [r50538-50571] Kevin P. Fleming <kpfleming@digium.com> - - * main/channel.c, main/pbx.c, include/asterisk/channel.h: make the - automatic post-answer delay happen only when the answer is - 'automatic' (not done by the Answer() dialplan application) - - * main/pbx.c, /: Merged revisions 50562 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r50562 | kpfleming | 2007-01-12 08:42:24 -0600 - (Fri, 12 Jan 2007) | 10 lines Merged revisions 50561 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 - Jan 2007) | 2 lines minor documentation clarification ........ - ................ - - * main/channel.c: when a channel gets automatically answered by an - application, sleep a bit to give the audio path (for VOIP - channels) time to be setup - -2007-01-11 05:54 +0000 [r50378-50469] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 50468 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 - lines Remove check for channel state as it can definitely be - something other then ring, and also clean up the code a bit. This - should solve the parking issues and maybe some attended transfer - issues people have been seeing. ........ - - * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: - Merged revisions 50466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 - lines Add support to see whether NAT was detected (yay symmetric - RTP) and also add a check in chan_sip so that if NAT has been - detected and the reinvite behind nat option has been turned off, - then just do partial bridge. (issue #8655 reported by mnicholson) - ........ - - * /, apps/app_speech_utils.c: Merged revisions 50433 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan - 2007) | 2 lines Merge speech-multi branch which adds support for - joining multiple sound files together to be played one after - another in SpeechBackground. ........ - - * /, main/config.c: Merged revisions 50405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50405 | file | 2007-01-10 14:46:29 -0500 (Wed, 10 Jan 2007) | 2 - lines Fix parsing when using something like ldap settings. (done - by anthonyl) ........ - - * include/asterisk/strings.h: Return the useless casts that ensure - this file is C++ clean. (issue #8602 reported by mikma) - - * /, channels/chan_sip.c: Merged revisions 50377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 - lines Fix chan_sip not working issue. Let's not prematurely - return 0. (issue #8783 reported by st41ker) ........ - -2007-01-10 16:47 +0000 [r50347] Jason Parker <jparker@digium.com> - - * /, cdr/cdr_manager.c: Merged revisions 50346 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50346 | qwell | 2007-01-10 10:45:36 -0600 (Wed, 10 Jan 2007) | 4 - lines Reverse some logic in cdr_manager, which made it fail to - load if the config file existed. Issue 8777 ........ - -2007-01-10 04:56 +0000 [r50267-50302] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 50298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r50298 | file | 2007-01-09 23:55:13 -0500 (Tue, - 09 Jan 2007) | 10 lines Merged revisions 50295 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 - lines Add another return value to dial_exec_full that indicates - execution is going to continuing at a new - extension/context/priority and to just let it slide. (issue #8598 - reported by jon) ........ ................ - - * channels/chan_zap.c: Allow usedistinctiveringdetection and - distinctiveringaftercid to be reset during a reload. (issue #8739 - reported by tzafrir) - - * main/pbx.c, /: Merged revisions 50266 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50266 | file | 2007-01-09 22:51:29 -0500 (Tue, 09 Jan 2007) | 2 - lines Ensure data's existence before trying to access it. (issue - #8774 reported by rcourtna) ........ - -2007-01-10 02:50 +0000 [r50229-50230] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Covert some spaces to tabs, and put a list - of defines in an enum. - - * Makefile, /: Merged revisions 50228 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r50228 | russell | 2007-01-09 21:17:46 -0500 - (Tue, 09 Jan 2007) | 14 lines Merged revisions 50227 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 - Jan 2007) | 6 lines Make the number that represents the major - version number a single digit instead of 2. Using two digits - makes it an octal number when put into version.h, which breaks - the compilation of any out of tree module that checks the version - for any version after 1.2.7 (reported by Matteo Brancaleoni on - the asterisk-dev mailing list, who gave credit to vihai for - pointing it out) ........ ................ - -2007-01-09 13:45 +0000 [r50152] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 50151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r50151 | tilghman | 2007-01-09 07:40:45 -0600 - (Tue, 09 Jan 2007) | 12 lines Merged revisions 50150 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 - Jan 2007) | 4 lines The advent of realtime has enabled people to - use commas in the fullname field. This could cause an issue with - sending voicemails, when the field is unquoted. (Issue 8595) - ........ ................ - -2007-01-09 12:25 +0000 [r50132] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Based on the following patch, changed for - trunk... Merged revisions 50124 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 - lines - handle re-invites properly in sip_hangup() - Add some - invitestate status changes just to be sure ........ - -2007-01-08 23:40 +0000 [r50099] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 50098 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50098 | qwell | 2007-01-08 17:39:12 -0600 (Mon, 08 Jan 2007) | 4 - lines Fix an issue with voicemail and users.conf, where it - wouldn't ever parse a password, since it was using "secret" - instead of "password" Issue 8761, reported by and patch - suggestion from ssokol. ........ - -2007-01-08 21:40 +0000 [r50075] Joshua Colp <jcolp@digium.com> - - * codecs/codec_zap.c: Move channel acquisition to when the - translation path is setup, and clean up. - -2007-01-08 21:17 +0000 [r50074] Matt O'Gorman <mogorman@digium.com> - - * /, apps/app_senddtmf.c: Merged revisions 50073 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ........ r50073 - | mogorman | 2007-01-08 15:11:16 -0600 (Mon, 08 Jan 2007) | 1 - line we can't unlock a channel if we cant find it. - AnthonyL bug - #8741 ........ - -2007-01-08 20:10 +0000 [r50033-50056] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Make callback declaration match one used in trunk. - - * include/asterisk/lock.h: Change trylock output for what already - has the lock from an error to a warning. - - * /, main/rtp.c: Merged revisions 50032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2 - lines Disable the more intense packet2packet bridging until the - bugs can be worked out. ........ - -2007-01-08 14:31 +0000 [r49931-50007] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 50006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 - lines Issue #8677 - Handle failure of T.38 re-invite This is not - a fix, but adding an error message to tell the admin that we have - a bad configuration. We should not send T.38 re-invites to - devices that can't handle it (with the current architecture where - you have to hard-code t.38 support per device). To really fix - this, we need to figure out a way to tell the incoming call that - the re-invite failed, so we can signal failure on that end and go - back to the original call. ........ - - * /, channels/chan_sip.c: Merged revisions 49983 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 - lines Issue #8524, support multiple via header values (tardieu) - Thanks! ........ - - * main/frame.c, include/asterisk/frame.h, main/rtp.c: Issue #8663 - - Add passthrough support for MPEG4 (neutrino88). - - * /, channels/chan_sip.c: Merged revisions 49945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 - lines We only need one forward declaration ........ - - * /, channels/chan_sip.c: Merged revisions 49925 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 - lines Issue 8735: Terminate state when extension is unavailable - for subscription ........ - -2007-01-08 05:13 +0000 [r49891] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 49890 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r49890 | file | 2007-01-08 00:11:54 -0500 (Mon, - 08 Jan 2007) | 10 lines Merged revisions 49889 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 - lines Ensure we use the default refresh value of 60 if the remote - server does not send one. (issue #8746 reported by maethor) - ........ ................ - -2007-01-08 03:56 +0000 [r49870] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, configure.ac: Merged revisions 49866 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r49866 | kpfleming | 2007-01-07 21:53:53 -0600 (Sun, 07 - Jan 2007) | 2 lines since we use AC_PATH_TOOL to find tools, we - should use the results it provides for us (reported by Brian - Capouch on the asterisk-dev list) ........ - -2007-01-07 21:46 +0000 [r49832-49835] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_dictate.c: Merged revisions 49834 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r49834 | tilghman | 2007-01-07 15:44:52 -0600 - (Sun, 07 Jan 2007) | 10 lines Merged revisions 49833 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 - Jan 2007) | 2 lines If openstream fails, then we crash (Issue - 8564) ........ ................ - - * /, channels/chan_sip.c: Merged revisions 49831 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) - | 2 lines Second condition was a subset of the first, so hold was - never decremented, thus hint stayed stuck (Issue 8747) ........ - -2007-01-07 19:00 +0000 [r49816] Joshua Colp <jcolp@digium.com> - - * funcs/func_base64.c, funcs/func_blacklist.c, - funcs/func_callerid.c: One const, two const. Let's stick with - everything else - one const. Plus older versions of GCC don't - support double const either. - -2007-01-07 16:21 +0000 [r49784-49801] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_odbc.c, include/asterisk/config.h, - res/res_realtime.c, main/config.c, funcs/func_realtime.c: When - calling the Realtime app more than once, unset fields which were - previously set are erroneously still set (Bug 6701). After - discussion, it was determined this should only be changed in - trunk. - - * funcs/func_shell.c, funcs/func_strings.c, funcs/func_cut.c: - Modifications to allow the output of SHELL() to be split per line - (Issue 8676) - - * funcs/func_shell.c (added): Add function to execute a shell - command and return the output (Issue 8676) - - * main/channel.c: Reduce duplication of code (Issue 6542) - -2007-01-07 07:43 +0000 [r49769] Jason Parker <jparker@digium.com> - - * main/indications.c: Fix a segfault when using "countries" that - don't have a matching zone. - -2007-01-06 00:28 +0000 [r49743] Jason Parker <jparker@digium.com> - - * main/pbx.c, /, res/res_features.c, pbx/pbx_config.c: Merged - revisions 49742 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 - lines Save 1 whopping byte of allocated memory! This looks like - it may have been a chicken/egg scenario.. You had to call a - cleanup func, because everything was allocated. Then since you - had to call a cleanup func, you were forced to allocate - ie; - strdup(""). ........ - -2007-01-06 00:13 +0000 [r49727-49741] Kevin P. Fleming <kpfleming@digium.com> - - * funcs/func_base64.c, funcs/func_rand.c, funcs/func_md5.c, - funcs/func_db.c, channels/chan_zap.c, funcs/func_module.c, - funcs/func_version.c, funcs/func_timeout.c, funcs/func_env.c, - funcs/func_strings.c, funcs/func_math.c, funcs/func_vmcount.c, - funcs/func_cut.c, include/asterisk/channel.h, funcs/func_sha1.c, - funcs/func_logic.c, funcs/func_uri.c, funcs/func_global.c, - funcs/func_realtime.c, funcs/func_enum.c, funcs/func_curl.c, - funcs/func_groupcount.c, funcs/func_odbc.c, - funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c, - funcs/func_callerid.c: finish const-ifying ast_func_read() - - * main/manager.c: probably shouldn't leave the mmap'ed file hanging - around in memory - - * /, configure, acinclude.m4: Merged revisions 49714-49715 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49714 | kpfleming | 2007-01-05 17:49:52 -0600 (Fri, 05 Jan 2007) - | 2 lines proper fix for r49712 ........ r49715 | kpfleming | - 2007-01-05 17:51:31 -0600 (Fri, 05 Jan 2007) | 2 lines one more - time... ........ - - * main/manager.c, include/asterisk/config.h, main/config.c: a - little more const-ification - -2007-01-05 23:51 +0000 [r49716] Joshua Colp <jcolp@digium.com> - - * codecs/codec_zap.c: It is possible for framein to get called and - no channel be available, so do a check before we increment the - count. - -2007-01-05 23:41 +0000 [r49711-49713] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, acinclude.m4: Merged revisions 49712 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r49712 | kpfleming | 2007-01-05 17:40:29 -0600 (Fri, 05 - Jan 2007) | 2 lines if --with-foo=<path> is specific for a - configure option, ensure that it is used for header file checking - as well ........ - - * main/pbx.c, /, channels/chan_sip.c, channels/chan_agent.c, - pbx/pbx_dundi.c, include/asterisk/pbx.h, apps/app_queue.c, - channels/chan_iax2.c, main/db.c, apps/app_speech_utils.c, - include/asterisk/astdb.h, apps/app_voicemail.c: const-ify some - more APIs, and fix rev 49710 from branch-1.4 in a better way here - -2007-01-05 23:31 +0000 [r49709] Matt O'Gorman <mogorman@digium.com> - - * codecs/codec_zap.c: no need to spam everyone with show transcoder - messages - -2007-01-05 23:17 +0000 [r49706] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /, codecs/codec_zap.c: Merged revisions - 49705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4 - lines Make codec_zap and chan_zap also depend on zaptel. This - fixes an issue (8727) with zaptel being in a different directory, - using --with-zaptel. ........ - -2007-01-05 22:53 +0000 [r49678-49681] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c, /: Merged revisions 49680 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49680 | kpfleming | 2007-01-05 16:52:37 -0600 (Fri, 05 Jan 2007) - | 2 lines don't 'consume' the params list before we try to use it - again ........ - - * main/manager.c: use mmap() to read in the results of the manager - action for an HTTP request, instead of reading it into a buffer - - * main/pbx.c, channels/chan_zap.c, /, channels/chan_sip.c, - apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, - utils/astman.c, res/res_jabber.c, include/asterisk/manager.h, - channels/chan_iax2.c, apps/app_queue.c, main/config.c, - res/res_monitor.c, main/manager.c, include/asterisk/jabber.h, - apps/app_senddtmf.c, main/db.c: Merged revisions 49676 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) - | 2 lines reduce stack consumption for AMI and AMI/HTTP requests - by nearly 20K in most cases ........ - -2007-01-05 22:18 +0000 [r49677] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 49675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49675 | file | 2007-01-05 17:14:47 -0500 (Fri, 05 Jan 2007) | 2 - lines Don't keep repeating the warning over and over when the end - of the call is reached. (issue #8724 reported by xrg) ........ - -2007-01-05 17:10 +0000 [r49578-49637] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_iax2.c: Merged revisions 49636 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r49636 | kpfleming | 2007-01-05 11:09:00 -0600 - (Fri, 05 Jan 2007) | 10 lines Merged revisions 49635 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 - Jan 2007) | 2 lines ensure that threads which are supposed to be - detached (because we aren't going to wait on them) are created - properly ........ ................ - - * main/threadstorage.c: use a rwlock-list for the list of TLS - objects - - * /, channels/chan_iax2.c: Merged revisions 49600 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007) - | 2 lines revert the dynamic_list insertion change... that was - not the right thing to do ........ - - * /, channels/chan_iax2.c: Merged revisions 49581 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007) - | 3 lines create the IAX2 processing threads as background - threads so they will use smaller stacks when we create a dynamic - thread, put it on the dynamic_list right away so we don't lose - track of it ........ - - * include/asterisk/strings.h: ensure that the proper - file/function/line shows up for dynamic string threadstorage - objects remove pointless casts - - * include/asterisk/threadstorage.h: yeah... so... compiling before - committing seems like it might be a good idea - - * build_tools/cflags.xml, include/asterisk.h, /, - main/threadstorage.c (added), main/Makefile, - include/asterisk/strings.h, include/asterisk/threadstorage.h, - main/asterisk.c: Merged revisions 49553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49553 | kpfleming | 2007-01-04 16:51:01 -0600 (Thu, 04 Jan 2007) - | 2 lines add support for tracking thread-local-storage objects - that exist via 'threadstorage' CLI commands ........ - -2007-01-04 23:02 +0000 [r49552-49573] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 49568 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2 - lines It's possible for the iax2 pvt to disappear, so if it - has... don't bother looking for dpentries. ........ - - * /, main/config.c: Merged revisions 49551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49551 | file | 2007-01-04 17:28:29 -0500 (Thu, 04 Jan 2007) | 2 - lines Only free comments and line buffer once we reach the first - level. (issue #8678 reported by ssokol, fixed by anthonyl) - ........ - -2007-01-04 21:59 +0000 [r49538] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c, /, channels/iax2-parser.c: Merged revisions 49536 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007) - | 2 lines don't mark these allocations as 'cache' allocations - when caching has been disabled ........ - -2007-01-04 21:40 +0000 [r49525] Joshua Colp <jcolp@digium.com> - - * main/manager.c: It's pretty difficult to pthread_kill a thread - that doesn't exist. (issue #8681 reported by bkruse) - -2007-01-04 21:06 +0000 [r49524] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/iax2-parser.c: Merged revisions 49523 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04 - Jan 2007) | 2 lines if we're going to decrement the frame count - when we free a frame, we should inrement it when we create one - :-) ........ - -2007-01-04 20:27 +0000 [r49491-49507] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: 1. Update osp guide. - - * configs/osp.conf.sample: 1. Update osp module configuration file. - -2007-01-04 18:32 +0000 [r49466] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2-parser.h, /, channels/chan_iax2.c, - channels/iax2-parser.c: Merged revisions 49465 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007) - | 2 lines only do IAX2 frame caching for voice and video frames - ........ - -2007-01-04 18:28 +0000 [r49464] Matt O'Gorman <mogorman@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 49459 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ................ - r49459 | mogorman | 2007-01-04 12:11:19 -0600 (Thu, 04 Jan 2007) - | 10 lines Merged revisions 49447 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 - | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 - lines converted a lot of 256 to PATH_MAX and some white space - fixes. ........ ................ - -2007-01-04 18:19 +0000 [r49463] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/Makefile, main/frame.c, /, channels/iax2-parser.c: Merged - revisions 49457,49460-49461 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) - | 2 lines make building of codec_gsm against the system GSM - library actually work ........ r49460 | kpfleming | 2007-01-04 - 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines don't define this - type either if LOW_MEMORY is enabled ........ r49461 | kpfleming - | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines don't do - frame header caching in the core if LOW_MEMORY is defined - ........ - -2007-01-04 18:17 +0000 [r49414-49462] Matt O'Gorman <mogorman@digium.com> - - * /, channels/iax2-parser.c: Merged revisions 49458 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.4 ........ - r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007) - | 2 lines don't do frame caching in LOW_MEMORY mode ........ - - * /, apps/app_voicemail.c: Merged revisions 49413 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ................ - r49413 | mogorman | 2007-01-04 10:50:56 -0600 (Thu, 04 Jan 2007) - | 11 lines Merged revisions 49412 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 - | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 - lines good catch russell sorry i missed that. fix magic number - with proper sizeof ........ ................ - -2007-01-03 23:41 +0000 [r49356] Matt O'Gorman <mogorman@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 49355 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ................ - r49355 | mogorman | 2007-01-03 17:32:03 -0600 (Wed, 03 Jan 2007) - | 14 lines Merged revisions 49354 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 - | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 - lines When using ODBC_STORAGE VoicemailMain doesn't create the - subdirectories for a mailbox such as the INBOX directory. this - patch solves that problem, was written by anthony be-125 ........ - ................ - -2007-01-03 11:15 +0000 [r49320-49321] Christian Richter <christian.richter@beronet.com> - - * doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, - /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - configs/misdn.conf.sample, channels/misdn/isdn_lib.c, - channels/misdn_config.c: Merged revisions 49313 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 - (Mi, 03 Jan 2007) | 41 lines Merged revisions - 48319,48321,48467,48552,48576,49135,49303 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | - 1 line changed a few debugs to higher debug levels ........ - r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | - 1 line added the export and import of the MISDN_ADDRESS_COMPLETE - Variable to inidcate wether the extension is already completely - dialed or if there might come additional digits by information - elements. also added some docs for that. ........ r48467 | - crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line - removed FIXUP state. added check for channel allocation conflict - when we create a setup while the other site creates a setup on - the same channel, besides the check we resolve this conflict. - ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 - Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a - preselected channel we just accept it, even when we're NT. added - some checks for segfaults. ........ r48576 | crichter | - 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we - reject a channel, because it's in use already, we shouldn't - process the setup anymore. made the channel allocation a bit - easier and more understandable, removed a few unused lines - ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 - Jan 2007) | 1 line added check for channel ranges in the - set/empty channel functions. set pmp_l1_check default to no. - added misdn restart pid cli command. added cleaning of channel - when we send a RELEASE_COMPLETE. ........ r49303 | crichter | - 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added - check for bridging in misdn_call to avoid setting - echocancellation when 2 mISDN channels are involved and when - bridging is set. That lead to a kernel panic before under - different situations, because we switched about 2 times between - hardware bridging and echocancelation * readded MISDN_URATE - variable which got lost before, this should make app_v110 work - again * fixed typo ........ ................ - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c, - channels/misdn_config.c: Merged revisions 47989 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47989 | crichter | 2006-11-24 16:46:13 +0100 - (Fr, 24 Nov 2006) | 9 lines Merged revisions 47968 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 - Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. - beatufied some logs, changed some loglevels. changed the default - value of block_on_alarm ........ ................ - -2007-01-03 03:28 +0000 [r49283] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /, Makefile.rules: Merged revisions 49282 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r49282 | kpfleming | 2007-01-02 21:21:25 -0600 (Tue, 02 - Jan 2007) | 2 lines various Makefile improvements to get chan_vpb - (and any other C++ modules) to build properly ........ - -2007-01-03 01:21 +0000 [r49260] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 49259 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 - lines Check pvt structure presence before passing to - send_command. This gets rid of the irritating message about a - packet without pvt structure. This happens because the scheduled - item is getting cancelled at almost the exact moment it is - getting executed. ........ - -2007-01-02 22:43 +0000 [r49238] Steve Murphy <murf@digium.com> - - * /, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex, - main/ast_expr2.fl: Merged revisions 49237 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49237 | murf | 2007-01-02 15:30:53 -0700 (Tue, 02 Jan 2007) | 1 - line This is a slight modification to Josh's edits for #8579; - both files edited were the produced by flex; so the source files - need to be changed instead, and the generated files regenerated. - ........ - -2007-01-02 20:02 +0000 [r49214-49215] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Removing propably accidentally added debug - messages sent to verbose channel - - * /, channels/chan_sip.c: Merged revisions 49212 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2 - lines Small cleanup of add_t38sdp - it's always enabled at that - point in the code ........ - -2007-01-02 17:04 +0000 [r49187] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_math.c: Tweak description text to match new - functionality (Issue 7959) - -2007-01-02 14:01 +0000 [r49166] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 49165 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007) - | 2 lines remove comment that is unrelated to this function - ........ - -2007-01-02 13:50 +0000 [r49152] Olle Johansson <oej@edvina.net> - - * /, configs/features.conf.sample: Update sample config - -2007-01-01 23:43 +0000 [r49100-49103] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /, build_tools/menuselect-deps.in, - configure, include/asterisk/autoconfig.h.in, configure.ac, - codecs/codec_zap.c: Merged revisions 49102 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) - | 2 lines check specifically for VLDTMF and transcoding support - in the system's Zaptel installation, and make only the modules - that need those features dependent on them (this will allow - building the other Zaptel-using parts of Asterisk against older - versions of Zaptel or those on other platforms that haven't - caught up yet to the Linux version) ........ - - * Makefile, sounds/Makefile: GNU make already knows what the - current directory is, there is no need to use 'pwd' - - * Makefile, /: Merged revisions 49098-49099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49098 | kpfleming | 2007-01-01 16:08:24 -0600 (Mon, 01 Jan 2007) - | 2 lines revert this change until a better solution can be - found... 'env -i' was not being used properly, but even when - changed to do so, this process fails during cross-compilation - because the menuselect build still sees 'CC' as set to the - cross-compiler ........ r49099 | kpfleming | 2007-01-01 16:48:03 - -0600 (Mon, 01 Jan 2007) | 2 lines use a simpler (and portable) - method to ensure that menuselect is built as a host binary - ........ - -2007-01-01 20:16 +0000 [r49092-49097] Olle Johansson <oej@edvina.net> - - * /: Block cleanup of release branch - - * include/asterisk/indications.h: Doxygen documentationification - - * main/manager.c: Fix manager too. - - * main/frame.c, channels/chan_sip.c, include/asterisk/frame.h: - - Add error handling to ast_parse_allow_disallow - Use this in - chan_sip configuration parsing - - * include/asterisk/acl.h, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, main/acl.c, - channels/chan_iax2.c, channels/chan_mgcp.c: - Implement error - handling in ast_append_ha - Use this in chan_sip - Document ha - functions in acl.c - -2006-12-31 19:15 +0000 [r49089] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: count is no longer used in the iaxq - structure really so let's just make this a statically declared - linked list. - -2006-12-31 09:38 +0000 [r49080-49082] Olle Johansson <oej@edvina.net> - - * CHANGES: Update CHANGES, make section about SIP. This might be a - good way to handle other parts of this file too, as it grows. - - * configs/sip.conf.sample: Added some docs - - * channels/chan_sip.c: Add version number to useragent string - - Issue #8700, blanchet - THANKS! - -2006-12-31 05:20 +0000 [r49075-49076] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_math.c: Add power and right/left shift functions - (Issue 7959) - - * configs/voicemail.conf.sample, UPGRADE.txt, apps/app_voicemail.c: - 1. Rename 'maxmessage' to 'maxsecs' to differentiate from - 'maxmsg'. 2. Rename 'minmessage' to 'minsecs' for parity. 3. Make - 'maxsecs' a per-user option, in addition to global. (Issue # - 8624) - -2006-12-30 18:32 +0000 [r49071-49074] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 49073 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49073 | file | 2006-12-30 13:31:17 -0500 (Sat, 30 Dec 2006) | 2 - lines IAX has been deprecated for quite some time so we had - better use IAX2 when creating the dial string for users. (issue - #8697 reported by ssokol) ........ - - * main/rtp.c: Clarify why we are reading in a frame in the - Packet2Packet bridge. - -2006-12-30 13:27 +0000 [r49068-49069] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: now that the 'languageprefix' option defaults to - 'on', and all channels have a default language of 'en', let's - install the English sound files into /var/lib/asterisk/sounds/en, - just like the other languages - - * main/channel.c: small formatting fix - -2006-12-30 05:49 +0000 [r49064-49067] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 49066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 - lines If the Packet2Packet bridge is being broken because of a - masquerade then attempt to read a frame in so the masquerade - actually happens. Otherwise weirdness will occur. (issue #8696 - reported by kjotte) ........ - - * funcs/func_odbc.c: Initialize obj pointers to NULL. Gets rid of - two compiler warnings. - - * /, channels/chan_iax2.c: Merged revisions 49063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2 - lines Initialize the packet queue in load_module instead of just - declaring the list with the default value. (issue #8695 reported - by ssokol) ........ - -2006-12-30 00:51 +0000 [r49062] Steve Murphy <murf@digium.com> - - * /, pbx/pbx_ael.c: Merged revisions 49061 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49061 | murf | 2006-12-29 17:40:37 -0700 (Fri, 29 Dec 2006) | 1 - line A fix for 8661, where the CUT func needed to have comma args - converted to vertical bars. I hope this change does little harm. - ........ - -2006-12-29 13:25 +0000 [r49056] Russell Bryant <russell@digium.com> - - * channels/chan_oss.c: Convert various comments to doxygen format. - -2006-12-29 11:02 +0000 [r49054] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Removing extra output - -2006-12-29 06:26 +0000 [r49053] Russell Bryant <russell@digium.com> - - * include/asterisk/smdi.h: Fix a spelling mistake in a comment. - -2006-12-29 00:33 +0000 [r49047] Kevin P. Fleming <kpfleming@digium.com> - - * /, BUGS: Merged revisions 49046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r49046 | kpfleming | 2006-12-28 18:32:59 -0600 - (Thu, 28 Dec 2006) | 10 lines Merged revisions 49045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 - Dec 2006) | 2 lines location of the bug posting guidelines has - changed ........ ................ - -2006-12-28 20:13 +0000 [r49030] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample, funcs/func_odbc.c, - funcs/func_strings.c: Integrate functionality tested on - svncommunity users back into trunk - -2006-12-28 20:10 +0000 [r49029] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/Makefile: Merged revisions 49028 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49028 | kpfleming | 2006-12-28 14:08:59 -0600 (Thu, 28 Dec 2006) - | 2 lines new versions of sounds ........ - -2006-12-28 20:05 +0000 [r49026-49027] Joshua Colp <jcolp@digium.com> - - * main/http.c: Convert uri_redirects list to read/write locks. - - * /, main/http.c: Merged revisions 49024 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49024 | qwell | 2006-12-28 14:52:46 -0500 (Thu, 28 Dec 2006) | 2 - lines make the uris_lock a rwlock instead of a mutex lock - needs - to be forward ported to trunk ........ - -2006-12-28 17:56 +0000 [r49019] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, - pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, - pbx/ael/ael.tab.h, utils/ael_main.c, main/ast_expr2.fl, - main/ast_expr2.c: Jason is having problems with the inclusion of - <err.h>; it appears to be unnecessary for sucessful builds, so I - either removed or commented out the inclusions from all the AEL - related code. New outputs from bison/flex are included, etc. - -2006-12-27 22:30 +0000 [r49010] Joshua Colp <jcolp@digium.com> - - * /, main/ast_expr2f.c, pbx/ael/ael_lex.c: Merged revisions 49009 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2 - lines ast_copy_string is not available when LOW_MEMORY is used - and things are being built in the utils directory, so we need to - resort to the old method of strncpy. (issue #8579 reported by - mottano) ........ - -2006-12-27 22:14 +0000 [r49007-49008] Kevin P. Fleming <kpfleming@digium.com> - - * main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, - main/dnsmgr.c, main/frame.c, main/manager.c, /, main/http.c, - main/logger.c, main/enum.c, main/asterisk.c, main/rtp.c, - main/term.c: Merged revisions 49006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) - | 2 lines since these variables all have static duration, none of - them need initializers (they default to zero anyway) ........ - - * codecs/g722: add file to ignore list - -2006-12-27 21:27 +0000 [r49004] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Only include include files once (imported - from 1.4) - -2006-12-27 21:21 +0000 [r48999-49001] Kevin P. Fleming <kpfleming@digium.com> - - * main/asterisk.c: apparently we need an explicit message to warn - people - - * main/file.c, UPGRADE.txt, main/asterisk.c, doc/asterisk-conf.txt: - make the 'languageprefix' option default to on, and deprecate - turning it off - - * /, main/file.c, include/asterisk/options.h, main/asterisk.c: - Merged revisions 48998 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) - | 3 lines move extern declaration for this option to a header - file where it belongs provide an initial value for - 'languageprefix' option, instead of relying on randomness to - provide a useful value ........ - -2006-12-27 20:30 +0000 [r48992-48996] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Only set "rfc2833compensate" option once - - * /, channels/chan_sip.c: Only handle T38 options once - - * channels/chan_sip.c: -Remove "localmask" setting (deprecated in - earlier version) - Remove "musiconhold" and "musicclass" settings - (also deprecated earlier) - -2006-12-27 18:34 +0000 [r48989-48990] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 48988 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) - | 2 lines make the option actually match the documentation - ........ - - * include/asterisk/utils.h, include/asterisk/astmm.h, main/frame.c, - /, main/astmm.c, channels/iax2-parser.c: Merged revisions 48987 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) - | 2 lines allow 'show memory' and 'show memory summary' to - distinguish memory allocations that were done for caching - purposes, so they don't look like memory leaks ........ - -2006-12-27 18:02 +0000 [r48976-48986] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Be politically - correct - - * apps/app_sms.c: From coding guidelines: Comments should explain - what the code does, not when something was changed or who changed - it. If you have done a larger contribution, make sure that you - are added to the CREDITS file. - - * /, channels/chan_sip.c, configs/sip.conf.sample: Add support for - buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid) - - * /, channels/chan_sip.c: Cleanup of handle_common_options - - * /, channels/chan_sip.c: Reset invitestate when sending new invite - - * /, channels/chan_sip.c: Issue #8600 - bogus SDP Content Length in - T.38 re-invite - -2006-12-26 05:23 +0000 [r48961-48967] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 48966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48966 | file | 2006-12-26 00:20:08 -0500 (Tue, 26 Dec 2006) | 2 - lines Get rid of a needless memory allocation and only create a - conference structure in find_conf_realtime if data was read from - realtime. (issue #8669 reported by robl) ........ - - * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: - Merged revisions 48964 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 - lines Add an API call that initializes an RTP structure. We need - this because chan_sip is cheeky and uses a temporary RTP - structure for codec purposes, and the API calls that are used - rely on the lock. (Pointed out on asterisk-dev by Andy Wang) - ........ - - * /, configure, configure.ac: Merged revisions 48960 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r48960 | file | 2006-12-25 12:04:48 -0500 (Mon, 25 Dec - 2006) | 2 lines Clean up autoconf file (gets rid of warnings seen - when rebuilding configure) and rebuild configure. ........ - -2006-12-25 06:42 +0000 [r48958-48959] Luigi Rizzo <rizzo@icir.org> - - * codecs/g722/g722.h: provide INT16_MIN and INT16_MAX for platforms - where they are not defined. - - * main/channel.c, apps/app_read.c, channels/chan_misdn.c, - funcs/func_channel.c, include/asterisk/indications.h, - apps/app_disa.c, main/app.c, res/snmp/agent.c, - contrib/utils/zones2indications.c, include/asterisk/channel.h, - res/res_indications.c, main/indications.c: rename the structs - struct tone_zone_sound and struct tone_zone defined in - indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid - conflicts with the structs with the same names defined in - tonezone.h Hope i haven't missed any instance. - -2006-12-25 05:22 +0000 [r48929-48957] Russell Bryant <russell@digium.com> - - * /, funcs/func_math.c: Merged revisions 48956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48956 | russell | 2006-12-25 00:21:20 -0500 - (Mon, 25 Dec 2006) | 14 lines Merged revisions 48955 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 - Dec 2006) | 6 lines Fix an error introduced by copying and - pasting the handling of the >= operator for the MATH function. If - a single equal sign was used as an operator, the function would - treat it is as if it were the >= operator. Now, it properly - handles it as an invalid operator. (issue #8665, patch by - tempest1) ........ ................ - - * funcs/func_callerid.c: Simplify the if statements used to check - to see if the argument was "num" or "number". It is not possible - to ever reach the second part of this conditional statement. - Thanks to my brother, Brett, for pointing this out. :) - - * main/frame.c: Resolve some compiler warnings - - * /, channels/chan_oss.c: Merged revisions 48948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) | - 3 lines Fix a typo in an error message that indicated that the - MGCP channel type could not be registered, instead of the correct - type, OSS. ........ - - * main/http.c, configs/http.conf.sample: Use spaces as a separator - for the redirect option to improve readability - - * /, channels/chan_iax2.c: Merged revisions 48944 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48944 | russell | 2006-12-24 02:25:38 -0500 - (Sun, 24 Dec 2006) | 11 lines Merged revisions 48943 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 - Dec 2006) | 3 lines Check for the proper return value on an error - in a call to mmap(). This was reported by Andy Wang on the - asterisk-dev list. Thanks! ........ ................ - - * channels/chan_sip.c: Merged revisions 48940 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48940 | russell | 2006-12-24 01:49:31 -0500 - (Sun, 24 Dec 2006) | 11 lines Merged revisions 48939 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 - Dec 2006) | 3 lines Remove a couple of misplaced dots in log - messages. This was reported by Andrea Spadaccini on the - asterisk-dev mailing list. ........ ................ - - * main/http.c: Simplify the definition of http_uri_redirect such - that only one allocation is done for exactly how much memory is - needed. This was suggested by Luigi on the asterisk-dev mailing - list. Thanks! - - * include/asterisk/http.h, main/http.c, CHANGES, - configs/http.conf.sample: - Convert the list of URI handlers to - use the linked list macros. While doing this, implementing - locking of this list to make it thread-safe. - Add a "redirect" - option to http.conf that allows redirecting one URI to another. I - was inspired to do this while playing with the Asterisk GUI. I - got tired of typing this URL to get to the GUI: - http://localhost:8088/asterisk/static/config/cfgadvanced.html So, - now I have the following line in http.conf: - redirect=/=/asterisk/static/config/cfgadvanced.html Now, I can - type the following into my browser and go to the GUI: - http://localhost:8088 - - * main/manager.c: Remove a debug message. If this is still needed - for debugging something, it should be made a LOG_DEBUG message. - -2006-12-23 19:55 +0000 [r48928] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: We should probably declare the lock... - and not just the constructor/deconstructor. - -2006-12-23 19:51 +0000 [r48927] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: Use the correct function to destroy an - rwlock in the destructor for an ast_rwlock_t - -2006-12-22 22:34 +0000 [r48871-48907] Jason Parker <jparker@digium.com> - - * Makefile, /, main/stdtime/localtime.c: Merged revisions 48906 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48906 | qwell | 2006-12-22 16:33:46 -0600 (Fri, 22 Dec 2006) | 2 - lines Minor fixes for Solaris. ........ - - * /, channels/chan_skinny.c: Merged revisions 48888 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec - 2006) | 2 lines Note to self: Run make before committing... - ........ - - * /, channels/chan_skinny.c: Merged revisions 48870 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec - 2006) | 2 lines Fix for issue 7774 - patch by alamantia ........ - -2006-12-22 10:35 +0000 [r48825-48857] Luigi Rizzo <rizzo@icir.org> - - * apps/app_sms.c: improve readability of a few macros. - - * apps/app_sms.c: make sms_hexdump() thread safe; restructure and - reduce indentation on some blocks. - - * apps/app_sms.c: make isodate thread-safe - - * apps/app_sms.c: - use the standard option parsing routines; - - document existing but undocumented parameters to send a message - (untested but unchanged; - ad a new option p(N) to set the - initial message delay to N ms so this can be adapted from the - dialplan to various countries; - -2006-12-21 21:57 +0000 [r48785-48817] Joshua Colp <jcolp@digium.com> - - * main/logger.c: Merge non-blocking logger from my branch. This - should improve things under heavy load with lots of CLI/logging - output. - - * /, redhat/asterisk.spec: Merged revisions 48783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48783 | file | 2006-12-21 15:26:29 -0500 (Thu, - 21 Dec 2006) | 10 lines Merged revisions 48782 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 - lines Add new silence sound files to the spec for Redhat. (issue - #8652 reported by alvaro_palma_aste) ........ ................ - -2006-12-21 20:15 +0000 [r48781] Steve Murphy <murf@digium.com> - - * codecs/codec_g722.c: This little mod gets rid of that g722 - compiler warning that breaks builds configured with - --enable-dev-mode; the previous commit of 48767 was to merge in - changes for bug 6334, unifying the open mode arguments for saner - operation. - -2006-12-21 19:52 +0000 [r48768] Luigi Rizzo <rizzo@icir.org> - - * apps/app_sms.c: put generator functions next to each other. - -2006-12-21 19:44 +0000 [r48767] Steve Murphy <murf@digium.com> - - * include/asterisk.h, channels/chan_zap.c, apps/app_meetme.c, - apps/app_festival.c, apps/app_dictate.c, apps/app_record.c, - res/res_convert.c, channels/chan_iax2.c, res/res_monitor.c, - cdr/cdr_sqlite.c, res/res_agi.c, main/file.c, main/app.c, - apps/app_sms.c, apps/app_directory.c, apps/app_chanspy.c, - apps/app_mixmonitor.c, main/db.c, apps/app_voicemail.c: a quick - fix to app_sms.c to get rid of cursed compiler warnings so I can - compile under --enable-dev-mode - -2006-12-21 19:36 +0000 [r48736-48766] Luigi Rizzo <rizzo@icir.org> - - * main/channel.c: same as in other places, check that - generator->release is not NULL before calling it. This allows - generators to set it to NULL when they have nothing to do there. - Later, the three copies of the code that releases a generator - should be moved to a function. - - * apps/app_sms.c: reduce indentation - - * apps/app_sms.c: restructure a block to reduce nesting - - * apps/app_sms.c: Add a bit of documentation on this code, - including pointers to relevant documents and comment on timing - issues. Initial merge of the code in - http://bugs.digium.com/view.php?id=8586 by Filippo Grassilli - (Hyppo) to support the SMS Protocol 2. In this commit i have - tried to minimize the diffs, so further code cleanup will come in - subsequent commits. - -2006-12-21 15:52 +0000 [r48723] Steve Murphy <murf@digium.com> - - * pbx/pbx_config.c: This small update will generate WARNINGS if - there is garbage in your extensions.conf file (liken extem => - instead of exten => !) - -2006-12-21 04:05 +0000 [r48680-48709] Joshua Colp <jcolp@digium.com> - - * include/asterisk/indications.h, main/indications.c: Really clean - up indications to use the linkedlists API - - * main/pbx.c: Switch list of global variables to read/write locks. - - * main/pbx.c: Convert alternate dialplan switch list to use - read/write locks. - -2006-12-21 00:24 +0000 [r48663] Steve Murphy <murf@digium.com> - - * configs/iax.conf.sample, main/jitterbuf.c, include/jitterbuf.h, - CHANGES, channels/chan_iax2.c: As per bug 7978, this version - introduces the jittertargetextra option in config files - -2006-12-21 00:11 +0000 [r48661-48662] Matthew Fredrickson <creslin@digium.com> - - * codecs/codec_g722.c: Minor addition giving props to Steve - Underwood for his hard work. Thanks again Steve! - - * codecs/Makefile, codecs/g722/Makefile (added), - codecs/codec_g722.c (added), codecs/g722/g722_encode.c (added), - codecs/g722 (added), build_tools/embed_modules.xml, - codecs/g722/g722_decode.c (added), codecs/g722/g722.h (added), - codecs/g722_slin_ex.h (added), codecs/slin_g722_ex.h (added): Add - codec G.722 support. - -2006-12-20 04:32 +0000 [r48638-48639] Joshua Colp <jcolp@digium.com> - - * apps/app_page.c: Clean up app_page - - * /, apps/app_voicemail.c: Merged revisions 48637 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48637 | file | 2006-12-19 21:56:09 -0500 (Tue, 19 Dec 2006) | 2 - lines vms doesn't exist on non-IMAP storage builds. ........ - -2006-12-20 00:13 +0000 [r48598-48599] Luigi Rizzo <rizzo@icir.org> - - * apps/app_sms.c: more formatting cleanup. Move some code into a - function sms_compose1() in preparation for supporting protocol 2 - as well. - - * apps/app_sms.c: formatting and code cleanup. Still a lot of - copy&pasted code here... - -2006-12-19 23:05 +0000 [r48591-48597] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48596 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48596 | file | 2006-12-19 18:04:30 -0500 (Tue, 19 Dec 2006) | 2 - lines Pass 'vms' pointer to record_and_review so it is then - passed to the IMAP store file function. (issue #8614 reported by - punknow) ........ - - * res/snmp/agent.c: Update res_snmp to use new API declaration of - pbx_builtin_serialize_variables (issue #8627 reported by - johann8384) - - * /, doc/snmp.txt: Merged revisions 48592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48592 | file | 2006-12-19 17:00:57 -0500 (Tue, 19 Dec 2006) | 2 - lines find is not the same as bind when it comes to - documentation. (issue #8626 reported by johann8384) ........ - - * res/res_limit.c: OpenBSD does not have RLIMIT_AS or RLIMIT_VMEM - so until someone finds the right rlimit to use then let's not - support the -v option on OpenBSD. (issue #8543 reported by jtodd) - -2006-12-19 21:32 +0000 [r48588-48589] Luigi Rizzo <rizzo@icir.org> - - * /: block 48583 - - * apps/app_sms.c: start documenting this code. On passing, fix the - bogus datalen on outgoing frames just fixed in 1.4 rev.48583 - -2006-12-19 21:28 +0000 [r48587] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/Makefile: Merged revisions 48586 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006) - | 2 lines suppress compiler warnings in this module until it can - be improved ........ - -2006-12-19 16:36 +0000 [r48580-48581] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dial.c: better name for struct dial_localuser. - - * main/cli.c: remove now useless extern declarations. - -2006-12-19 14:57 +0000 [r48578] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, /, funcs/func_odbc.c: Merged revisions - 48577 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48577 | kpfleming | 2006-12-19 08:57:09 -0600 (Tue, 19 Dec 2006) - | 2 lines use the proper variable type for these unixODBC API - calls, eliminating warnings on 64-bit platforms that use the - 'new' 64-bit types for ODBC API calls ........ - -2006-12-19 09:58 +0000 [r48573-48575] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dial.c: introduce a temporary variable for tmp->chan to - shorten expressions. - - * apps/app_dial.c: stop what i think is a memory leak in case Dial - fails to connect to a channel. Before committing to 1.4 i would - like some other people to review and test this fix - thanks. - - * apps/app_dial.c: move a large block related to privacy handling - to a separate function. - -2006-12-19 03:47 +0000 [r48572] Joshua Colp <jcolp@digium.com> - - * Makefile, /: Merged revisions 48571 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48571 | file | 2006-12-18 22:46:12 -0500 (Mon, 18 Dec 2006) | 2 - lines Use env -i to start a fresh environment when going to build - menuselect. This is more portable then using unset. (issue #8543 - reported by jtodd) ........ - -2006-12-18 17:44 +0000 [r48568] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/channel.h: unbreak the macro used for - incrementing the frame counters. I don't know when the bug was - introduced, but with the typical usage c->fin = - FRAMECOUNT_INC(c->fin) the frame counters stay to 0. - -2006-12-18 17:30 +0000 [r48565-48567] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Clean up find_idle_thread function and use - atomic operations for dynamic thread count. - - * /, channels/chan_iax2.c: Merged revisions 48564 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 - lines Put thread into proper list if we abort handling due to an - error, and also hold the lock while putting it back into the - proper idle list so we don't prematurely get a signal. (issue - #8604 reported by arkadia) ........ - -2006-12-18 16:57 +0000 [r48562-48563] Jason Parker <jparker@digium.com> - - * configure.ac: ctrl-w != w (nano search) (surprisingly, the fix - was ever so slightly different in 1.4) - -2006-12-18 16:24 +0000 [r48558-48560] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/strings.h: apply the proposed fix for bug 8602 - http://bugs.digium.com/view.php?id=8602 (i am not sure if there - is still missing cast in front of the alloca() call - being a - macro this is probably triggered only when actually used). Add - function ast_str_reset() to reinitialize the string to an empty - string instead of playing with the internal fields. - - * main/cdr.c, main/pbx.c, apps/app_dumpchan.c, - include/asterisk/cdr.h, include/asterisk/pbx.h, apps/app_queue.c, - main/cli.c: convert the final clients of ast_build_string to use - ast_str_*() Now the only module left using it is chan_sip.c - - * main/logger.c: debugging shows that we always need more than 128 - bytes for the verbose and logging messages so start with a larger - buffer from the beginning. - -2006-12-18 11:59 +0000 [r48555] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/Makefile, codecs/gsm/Makefile, utils/astman.c, - utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c, - codecs/lpc10/Makefile: Merged revisions 48554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006) - | 3 lines remove some now-unnecessary explicit includes of - autoconfig.h clean up per-file dependencies during 'make clean' - ........ - -2006-12-18 11:28 +0000 [r48550-48553] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: Replace ast_build_string with ast_str_*(). On - passing remove presumably duplicate code to generate the message - for the manager_hooks: in the previous version, the message was - almost the same as the one sent to regular sessions, with the - exception of the empty line at the end, and a few (presumably - unintentional) differences e.g. timestamps, debugging, and - lowercase headers for "event" and "privilege". now we reuse the - same message as before. - - * funcs/func_realtime.c: replace ast_build_string() with - ast_str_*(). Unless i am very mistaken, function_realtime_read() - was broken in that it would always return an empty string - (because ast_build_string() advanced the pointer to the end of - the string, and there was no reference to the initial value. This - commit should fix this problem. - - * apps/app_queue.c: replace ast_build_string() with ast_str_*(); - simplify __queues_show() - -2006-12-17 18:33 +0000 [r48549] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/prep_tarball: Merged revisions 48548 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r48548 | kpfleming | 2006-12-17 12:33:24 -0600 (Sun, 17 - Dec 2006) | 2 lines need an additional argument here to make the - downloads actually occur ........ - -2006-12-17 12:47 +0000 [r48543] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: define a mask SIP_INSECURE sam as for other - sets of flags. - -2006-12-16 21:38 +0000 [r48522-48529] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: Merged revisions 48528 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48528 | kpfleming | 2006-12-16 15:34:41 -0600 (Sat, 16 Dec 2006) - | 2 lines use m4 quoting for AC_MSG_NOTICE calls, to keep these - calls from thinking they have multiple arguments ........ - - * /, agi, codecs, utils, main/Makefile, apps, - Makefile.moddir_rules, Makefile.rules, cdr, codecs/ilbc, formats, - utils/Makefile, agi/Makefile, Makefile, funcs, main/db1-ast, - codecs/lpc10, build_tools/mkdep (removed), main, codecs/gsm, res, - pbx, channels: Merged revisions 48525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006) - | 2 lines simplify dependency tracking system, using the - compiler's built-in method for generating them, and only doing - dependency tracking if developer mode is enabled via the - configure script ........ - - * funcs/func_curl.c: update to use trunk's version of the - threadstorage API - - * Makefile, include/asterisk.h, /, main/stdtime/localtime.c: Merged - revisions 48521 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) - | 2 lines since we really, really have to have autoconfig.h - included before all other headers (especially system headers), - the Makefile will now force it to happen (this will fix build - problems with files like ast_expr2f.c, where we can't control the - inclusion order in the file itself) ........ - -2006-12-16 11:23 +0000 [r48515-48520] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c: forgot this part... - - * main/cli.c: another conversion from ast_build_str to ast_str - - * main/translate.c: convert ast_build_str to ast_str_* - - * include/asterisk/http.h, main/manager.c, main/http.c, - include/asterisk/strings.h: replace ast_build_string() with - ast_str_*() functions. This makes the code easier to follow and - saves some copies to intermediate buffers. - -2006-12-16 04:25 +0000 [r48514] Kevin P. Fleming <kpfleming@digium.com> - - * funcs/func_curl.c, /: Merged revisions 48513 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48513 | kpfleming | 2006-12-15 22:25:09 -0600 (Fri, 15 Dec 2006) - | 2 lines instead of initializing the curl library every time the - CURL() function is invoked, do it only once per thread (this - allows multiple calls to CURL() in the dialplan for a channel to - run much more quickly, and also to re-use connections to the - server) (thanks to JerJer for frequently complaining about this - performance problem) ........ - -2006-12-16 02:42 +0000 [r48508-48512] Luigi Rizzo <rizzo@icir.org> - - * res/res_limit.c: prevent a compiler warning - - * main/manager.c, main/logger.c, main/utils.c, - include/asterisk/strings.h, main/cli.c: simplify the - ast_dynamic_str_*.... routines by renaming them to ast_str ... - and putting the struct ast_threadstorage pointer into the struct - ast_str. This makes the code a lot more readable. At this point - we can use these routines also to replace ast_build_string(). - - * include/asterisk/utils.h, main/utils.c, - include/asterisk/strings.h, include/asterisk/threadstorage.h: - move the dynamic string support in a better place i.e. string.h - While doing this, add a bit of documentation, and slightly extend - the functionality as follows: + a max_len of -1 means that we - take whatever the current size is, and never try to extend the - buffer; + add support for alloca()-ted dynamic strings, which is - very useful for all cases where we do an ast_build_string() now. - Next step is to simplify the interface by using shorter names - (e.g. ast_str as a prefix) and removing the _thread variant of - the functions by saving the threadstorage reference into the - struct ast_str. This can be done by overloading the 'type' field. - Finally, I will do my best to remove the convoluted interface - that results from trying to support platforms without va_copy(). - - * res/res_smdi.c: remove a duplicate include - -2006-12-15 19:57 +0000 [r48503-48507] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 48506 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2 - lines Turn payload_lock into bridge_lock and make it encompass - all RTP structure contents that may relate to bridge information, - including who we are bridged to. ........ - - * /, channels/chan_iax2.c: Merged revisions 48504 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2 - lines Hold call structure lock in places where a qualify or peer - action can destroy it. ........ - - * /, channels/chan_iax2.c: Merged revisions 48502 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2 - lines Lock network retransmission queue in all places that it is - used. ........ - -2006-12-15 18:37 +0000 [r48495-48501] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: unbreak the output for http session. Not long ago - i replaced lseek() with fseek() but forgot that filr FILE's you - need ftell to give you the current position. - - * main/channel.c, include/asterisk/channel.h: remove - ast_safe_string_alloc() - it is completely equivalent to - asprintf(). - - * channels/chan_zap.c: replace ast_safe_string_alloc() with - asprintf() - - * channels/chan_features.c: replace ast_safe_string_alloc() with - asprintf() - - * include/asterisk/threadstorage.h: small documentation - improvements. - -2006-12-15 13:36 +0000 [r48485-48491] Olle Johansson <oej@edvina.net> - - * main/tdd.c, include/asterisk/tdd.h: Doxygen changes - - * /, channels/chan_sip.c: Issue #8592 - treat 504 as congestion - (imported from 1.2/1.4) - - * /, channels/chan_sip.c: Update to latest IANA specs - -2006-12-15 06:34 +0000 [r48479-48480] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: Add support to see what holds the lock - when doing trylock. - - * /, channels/chan_iax2.c: Merged revisions 48478 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2 - lines Use a wakeup variable so that we don't wait on IO - indefinitely if packets need to be retransmitted. ........ - -2006-12-15 04:03 +0000 [r48476-48477] Luigi Rizzo <rizzo@icir.org> - - * main/channel.c, include/asterisk/channel.h: constify - ast_state2str() and note it is not reentrant. - - * main/pbx.c, include/asterisk/channel.h: remove the macro LOAD_OH - and expand it inline in the only place where it was used. - -2006-12-14 17:39 +0000 [r48462-48473] Joshua Colp <jcolp@digium.com> - - * /, include/asterisk/rtp.h, main/rtp.c: Merged revisions 48472 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 - lines Payload values on the RTP structure can change AFTER a - bridge has started. This comes from the packet handling of the - SIP response when indication that it was answered has been sent. - Therefore we need to protect this data with a lock when we - read/write. (issue #8232 reported by tgrman) ........ - - * /, main/rtp.c: Merged revisions 48461 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2 - lines Remove direct RTCP bridging. I've come to the conclusion - that we should handle this through the core and not just forward - it on. Should solve a few bugs. ........ - -2006-12-13 23:08 +0000 [r48458-48459] Luigi Rizzo <rizzo@icir.org> - - * main/pbx.c: make sure that showdialplan sends only one 'Response: - Success ' message even in case of a recursive call. - - * main/pbx.c: clean up function manager_show_dialplan_helper() - reducing indentation and normalizing loops. While doing this, - remove some unused variables, fix an uninitialized string - (idaction), and mark some places where the behaviour is not what - we would expect (e.g. an empty context is reported as an error - same as a non-existent one). Given that this function is not in - 1.4, the above can be changed without too many backward - compatibility concerns. Not applicable to 1.4 or below. - -2006-12-13 21:23 +0000 [r48455] Matt O'Gorman <mogorman@digium.com> - - * codecs/codec_zap.c: support for deactivating translation paths - that are no longer available and more descriptive show transcoder - cli command. - -2006-12-13 00:56 +0000 [r48433] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: revert check for a zaptel transcoder related - definition that was done in the wrong module. - -2006-12-12 23:28 +0000 [r48432] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/prep_tarball: Merged revisions 48427 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r48427 | kpfleming | 2006-12-12 17:18:14 -0600 (Tue, 12 - Dec 2006) | 2 lines when making a release, we can always use wget - and we can't run the configure script to find that out... - ........ - -2006-12-12 22:32 +0000 [r48416-48417] Russell Bryant <russell@digium.com> - - * include/asterisk/app.h, channels/chan_sip.c, - include/asterisk/channel.h, include/asterisk/pbx.h: Fix various - spelling mistakes in comments. - - * channels/chan_zap.c: Make chan_zap inform you that your version - of zaptel is too old instead of just failing to compile. It seems - like the proper way to do this would be in the configure script. - However, that wouldn't help existing checkouts unless we forced - the configure script to be executed after any code was changed. - -2006-12-12 19:55 +0000 [r48415] Matt O'Gorman <mogorman@digium.com> - - * codecs/codec_zap.c: fixed nubb error on my part, transcoder now - unlocks and locks correctly, as well as counts in the correct - direction. - -2006-12-12 10:36 +0000 [r48408-48410] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: properly initialize a malloc'ed buffer - - * main/manager.c: normalize the scanning of "general" options in - the config file. - - * main/cli.c: Make sure tab-completion works even when we have - typed a fully matching word (e.g. "sip<TAB>"); this is - implemented by this one-line change - for (;; dst++, src += n) { - + for (;src < argindex; dst++, src += n) { However this code is - not exactly trivial to understand, so i am also adding some - comments to help figuring out what it does. - -2006-12-12 04:14 +0000 [r48402] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48401 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48401 | file | 2006-12-11 23:13:48 -0500 (Mon, 11 Dec 2006) | 2 - lines Use S_OR in my previous app_voicemail. This is the way it - should have been done. ........ - -2006-12-11 23:02 +0000 [r48397-48400] Matt O'Gorman <mogorman@digium.com> - - * /, sounds/Makefile: Merged revisions 48399 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ........ r48399 - | mogorman | 2006-12-11 17:02:10 -0600 (Mon, 11 Dec 2006) | 2 - lines new sounds package with 100% more silence ........ - - * /, apps/app_externalivr.c: Merged revisions 48396 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.4 - ................ r48396 | mogorman | 2006-12-11 16:11:35 -0600 - (Mon, 11 Dec 2006) | 12 lines Merged revisions 48394 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.2 ........ - r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) - | 4 lines app_externalivr needs a real silence file, and - additional changes to add silence files into core instead of - extra patch provided by bug 8177 with minor additions. ........ - ................ - -2006-12-11 21:35 +0000 [r48392] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48391 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48391 | file | 2006-12-11 16:31:23 -0500 (Mon, 11 Dec 2006) | 2 - lines Return non-existant callerid handling to that which it was - before. In 1.4 and trunk callerid can be allocated but not have - any contents so we have to use ast_strlen_zero before passing it - to the relevant functions. (issue #8567 reported by pabelanger) - ........ - -2006-12-11 21:04 +0000 [r48390] Matt O'Gorman <mogorman@digium.com> - - * codecs/codec_zap.c: add support for dynamic channel creation and - destruction, and show transcoder to show number of channels in - use. - -2006-12-11 18:11 +0000 [r48389] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: make sure the argument to ast_malloc() is > 0. - Long explaination: The behaviour of the underlying malloc(0) - differs depending on the operating system. Some return NULL (SysV - behaviour); some still allocate a small chunk of memory and - return a valid pointer (e.g. traditional BSD); some (e.g. FreeBSD - 6.x) return a non-null pointer that causes a memory fault if - used, even just for reading. Given the above variety, better - never call malloc(0). - -2006-12-11 17:00 +0000 [r48388] Steve Murphy <murf@digium.com> - - * main/app.c: This update fixes the problem reported in bug 8551; - that ast_app_getdata() behaves differently in trunk for the case - of a null prompt. - -2006-12-11 05:40 +0000 [r48384] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 48382 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48382 | tilghman | 2006-12-10 23:37:09 -0600 (Sun, 10 Dec 2006) - | 2 lines STRFTIME() does not actually require an argument (issue - 8540) ........ - -2006-12-11 05:38 +0000 [r48378-48383] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 48381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2 - lines Merge in my latest RTP changes. Break out RTP and RTCP - callback functions so they no longer share a common one. ........ - - * /, apps/app_meetme.c: Merged revisions 48379 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48379 | file | 2006-12-11 00:30:01 -0500 (Mon, 11 Dec 2006) | 2 - lines Use the correct API call to say a device state changed. - (Yes, I'm a nub.) ........ - - * /, apps/app_meetme.c: Merged revisions 48377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48377 | file | 2006-12-10 23:57:38 -0500 (Sun, 10 Dec 2006) | 2 - lines Don't access the conference structure after it has been - freed. ........ - -2006-12-11 00:52 +0000 [r48376] Tilghman Lesher <tlesher@digium.com> - - * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, - res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, - apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48375 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48375 | tilghman | 2006-12-10 18:47:21 -0600 - (Sun, 10 Dec 2006) | 13 lines Merged revisions 48374 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 - Dec 2006) | 5 lines When doing a fork() and exec(), two problems - existed (Issue 8086): 1) Ignored signals stayed ignored after the - exec(). 2) Signals could possibly fire between the fork() and - exec(), causing Asterisk signal handlers within the child to - execute, which caused nasty race conditions. ........ - ................ - -2006-12-10 03:14 +0000 [r48373] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 48372 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat, - 09 Dec 2006) | 9 lines Merged revisions 48371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 - line This version applies the patch suggested by stevens in bug - 7836 (make inbound channel RINGING state consistent with other - channels). ........ ................ - -2006-12-09 16:44 +0000 [r48359-48365] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: convert the thread IO state and type to use - enums. - - * /, channels/chan_iax2.c: Merged revisions 48363 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | - 8 lines Use locking when accessing the registrations list. This - list is not actually used very often, so the likelihood of there - being a problem is pretty small, but still possible. For example, - if the CLI command to list the registrations was called at the - same time that a reload was occurring and the registrations list - was getting destroyed and rebuilt, a crash could occur. In - passing, go ahead and convert this list to use the linked list - macros. ........ - - * channels/chan_iax2.c: chan_iax2 has an extremely large function, - socket_process(), to handle incoming frames. The function, before - this commit, was roughly 1400 lines long. So, I am working on - breaking this up into functions so that the code is easier to - follow and debug. Also, I will be committing these changes in - chunks as I do them to ease tracking down any potentially - introduced problems. Break out roughly 150 lines from - socket_process() and introduce a new function, - socket_process_meta() which handles the parsing of an incoming - meta frame. Also, restructure some of this code a bit to reduce - the deep nesting that was in this code. - - * channels/chan_iax2.c: - Fix a few spelling mistakes - Use - sizeof() to pass an array size to a function - Use a single bit - for a variable in the chan_iax2_pvt stuct since that is all it - needs. - Add some comments about the iaxs, iaxl, and lastused - arrays. - -2006-12-07 18:21 +0000 [r48358] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 48357 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48357 | russell | 2006-12-07 13:17:28 -0500 - (Thu, 07 Dec 2006) | 11 lines Merged revisions 48356 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 - Dec 2006) | 3 lines Ensure that the file position is not - incremented beyond the total number of files available for - playback. (issue #8539, ulogic) ........ ................ - -2006-12-07 16:42 +0000 [r48351] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/http.h, main/manager.c, main/http.c, - configs/manager.conf.sample: - Generalize the function - ssl_setup() so that the certificate info are passed as an - argument. - Update the code in main/http.c to use the new - interface (the diff is large but mostly mechanical, due to the - name change of several variables); - And since now it is trivial, - implement "AMI over TLS", and document the possible options in - manager.conf - And since the test client (openssl s_client - -connect host:port ) does not generate \r\n as a line terminator, - make get_input() also accept just a \n as a line terminator (Mac - users: do you also need the \r-only version ?) The option parsing - in manager.conf is not very efficient, and needs to be cleaned up - and made similar to what we have in http.conf - -2006-12-07 16:03 +0000 [r48350] Steve Murphy <murf@digium.com> - - * main/manager.c, /: Merged revisions - 47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri, - 24 Nov 2006) | 6 lines Doxygen update - Document cause codes - - Document a bit more on channel variables - global, predefined and - local - Fix some doxygen in channel.h. Adding one comment for two - definitions does not work. They won't be copied to each. - ................ r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri, - 24 Nov 2006) | 1 line This fix inspired by a patch supplied in - bug 8189, which points out problems with the PLC code - ................ r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri, - 24 Nov 2006) | 1 line removed the svnmerge-integrated property - from trunk; it's confusing svnmerge in newly created branches - ................ r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat, - 25 Nov 2006) | 5 lines set pointers to NULL after freeing memory - to avoid multiple free() probably 1.4/1.2 issue as well if - someone can look into that. ................ r48003 | oej | - 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines - Adding - comment on suspicious memory allocation. Seems like it's never - freed, but I don't have a clear understanding of the frame - allocation/deallocation, so I just mark this for investigation. - (Reported by Ed Guy). We're trying to see if a free() hurts... - - Doxygen comments on p2p rtp bridge stuff. I am a bit worried - about shortcutting rtcp this way, but will need feedback from - rtcp gurus. This should work for video calls too, and possibly - UDPTL. ................ r48004 | oej | 2006-11-25 02:48:30 -0700 - (Sat, 25 Nov 2006) | 2 lines Changing ERROR to lesser level. - Imported from 1.2/1.4 ................ r48008 | rizzo | - 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines generalize - a bit the functions used to create an tcp socket and then run a - service on it. The code in manager.c does essentially the same - things, so we will be able to reuse the code in here (probably - moving it to netsock.c or another appropriate library file). - ................ r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat, - 25 Nov 2006) | 1 line Updates to show linkset command - ................ r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat, - 25 Nov 2006) | 2 lines Add ss7 show linkset command - ................ r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat, - 25 Nov 2006) | 1 line Make sure we don't send a group reset on a - group larger than 32 CICs ................ r48012 | mattf | - 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line bug fix - ................ r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat, - 25 Nov 2006) | 1 line Make compiler happier ................ - r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1 - line Little fix so we use the right message ................ - r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9 - lines Merged revisions 48015 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 - line A little bit of func_cdr documentation upgrade-- no bug# - involved, although 8221 may have inspired it. ........ - ................ r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat, - 25 Nov 2006) | 9 lines Merged revisions 48017 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 - line might as well also document the raw values of the flag vars - ........ ................ r48019 | russell | 2006-11-25 23:55:33 - -0700 (Sat, 25 Nov 2006) | 6 lines - Add some comments on thread - storage with a brief explanation of what it is as well as what - the motivation is for using it. - Add a comment by the - declaration of ast_inet_ntoa() noting that this function is not - reentrant, and the result of a previous call to the function is - no longer valid after calling it again. ................ - -2006-12-06 20:46 +0000 [r48332-48338] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: remove duplicated code to start the server - threads, use the infrastructure exposed in http.c earlier today. - As a bonus, now we can restart the session on a different port - just reloading the module. On passing, fix a bug in the handling - of 'enabled' in the configuration file - previously, a missing - "enabled=" line in manager.conf meant "whatever the state was - before" instead of a specific value. - - * main/manager.c: Part of the transformations necessary to add TLS - support, as described in - http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html - In detail, this commit does the following: b) change the function - get_input() to use fread() instead of read() to collect the data. - One can still do the ast_wait_for_input() on the original - descriptor returned by accept(). c) change the function - send_string() to work on the FILE *. As a side effect, this - change now really guarantees that we don't spend more than - "writetimeout" milliseconds on each line sent. d) modify the - function action_command() so that it creates a temporary file - descriptor to be passed to ast_cli_command(), and then read back - the data from the temp file and write it to the output with - send_string(). The code is similar to what is done in - generic_http_callback() to support AMI-over-HTTP. - -2006-12-06 16:54 +0000 [r48327] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Handle multiple 487's correctly - -2006-12-06 16:19 +0000 [r48325] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample, /: Merged revisions 48323 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48323 | russell | 2006-12-06 11:15:45 -0500 - (Wed, 06 Dec 2006) | 11 lines Merged revisions 48322 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 - Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option - in the sample configuration file. (issue #8526, arkadia) ........ - ................ - -2006-12-06 16:17 +0000 [r48324] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/http.h, main/http.c: Make externally visible - some generic code useful to create and implement services over - tcp and/or tcp-tls. This commit is nothing more than moving - structure definitions (and documentation) from main/http.c to - include/asterisk/http.h (temporary location until we find a - better place), and removing the 'static' qualifier from - server_root() and server_start(). The name change (adding the - ast_ prefix as a minimum, and then possibly a more meaningful - name) is postponed to future commits. Does not apply to other - versions of asterisk. - -2006-12-06 12:34 +0000 [r48318] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't send Contact in SIP Messages - (imported from 1.2/1.4). Reported by Gunnar at Omnitor. - -2006-12-06 07:39 +0000 [r48299-48307] Russell Bryant <russell@digium.com> - - * apps/app_osplookup.c, apps/app_meetme.c, apps/app_queue.c, - apps/app_voicemail.c: Resolve some pointer signedness compiler - warnings in app_osplookup, and constify a bunch of usage strings - for CLI commands. - - * channels/chan_local.c, channels/chan_skinny.c, - channels/chan_agent.c, channels/chan_features.c, - channels/chan_alsa.c, channels/iax2-provision.c, - channels/chan_gtalk.c, channels/chan_oss.c, channels/chan_mgcp.c: - Constify a bunch of usage strings for CLI commands. - - * res/res_config_pgsql.c, res/res_limit.c, res/res_agi.c, - res/res_crypto.c, res/res_realtime.c, res/res_jabber.c, - res/res_odbc.c: Constify a bunch of usage strings for CLI - commands. - - * main/channel.c, main/udptl.c, main/frame.c, main/translate.c, - main/file.c, pbx/pbx_dundi.c, main/db.c, main/rtp.c: Staticize - one, and Constify a bunch of usage strings for CLI commands. - - * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, - main/asterisk.c, main/cli.c: Constify a bunch of the usage - strings for CLI commands. - - * channels/chan_iax2.c: Instead of creating an unused instance of - an unnamed enum, give it a name. - - * include/asterisk/cli.h: Make the "usage" member of the - ast_cli_entry struct const to resolve a compiler warning. - -2006-12-05 20:46 +0000 [r48282] Joshua Colp <jcolp@digium.com> - - * configure: Regenerate configure for Qwell's last commit. - -2006-12-05 20:44 +0000 [r48280] Jason Parker <jparker@digium.com> - - * /, configure.ac: Merged revisions 48279 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48279 | qwell | 2006-12-05 14:42:52 -0600 (Tue, 05 Dec 2006) | 4 - lines Fix curl version number testing to be much more friendly to - non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX - compliant now.. ........ - -2006-12-05 20:39 +0000 [r48277] Olle Johansson <oej@edvina.net> - - * include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c: - Doxygen updates - -2006-12-05 20:15 +0000 [r48276] Jason Parker <jparker@digium.com> - - * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c, - main/fskmodem.c: Expand on r48273 (from issue 8506), to translate - more of the fskmodem stuff to English. r48273 dealt with the - comments and such, this deals with the code itself. (This - couldn't have been easily done if it weren't for 48273 - thanks - again for that merbanan) - -2006-12-05 19:41 +0000 [r48269-48273] Olle Johansson <oej@edvina.net> - - * include/asterisk/fskmodem.h, main/fskmodem.c: Issue #8506 - - translate spanish comments in fskmodem to english (according to - bug guidelines) Thanks merbanan! - - * /: Blocking invitestate patch that is already merged to svn - trunk. - - * /, configs/sip.conf.sample: Adding docs on t.38 - -2006-12-05 14:33 +0000 [r48266] TransNexus OSP Development <support@transnexus.com> - - * apps/app_osplookup.c: 1. Change to remove the compiling warning: - "app_osplookup.c:2169: warning: initialization discards - qualifiers from pointer target type" - -2006-12-05 11:09 +0000 [r48258-48259] Olle Johansson <oej@edvina.net> - - * main/frame.c, include/asterisk/frame.h, main/rtp.c: Well, yes... - - * main/frame.c, include/asterisk/frame.h, main/rtp.c: Reserving - flags for coming code (currently in the "videocaps" branch) - implementing T.140 support in RTP. T.140/RFC 4351 is TDD over IP - - text telephony for hearing impaired. It defines a realtime text - chat, much like the old "talk" application in Unix. T.140 is - character by character in real time. It's not the same as our - current MESSAGE format - that is more like IM, but can be - gatewayed to MESSAGE with a text "codec" if needed. More patches - will follow, as soon as we've separated this code from the video - capabilities functions in the videocaps branch. Code by John - Martin, Aupix (disclaimer on file) - -2006-12-05 01:46 +0000 [r48253-48255] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48254 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48254 | tilghman | 2006-12-04 19:41:02 -0600 (Mon, 04 Dec 2006) - | 2 lines Oops, forgot to release the odbc handle ........ - - * /, apps/app_voicemail.c: Merged revisions 48252 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48252 | tilghman | 2006-12-04 19:34:34 -0600 - (Mon, 04 Dec 2006) | 14 lines Merged revisions 48251 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 - Dec 2006) | 6 lines If the recording in the database is too - large, it will fail to retrieve with an mmap error. Not too sure - why this doesn't happen when we put it in the database, also, but - since that doesn't seem to be broken, I'm not going to fix it (at - least until someone reports it). Solution is to ask for the file - in smaller chunks. (Bug 8385) ........ ................ - -2006-12-04 21:49 +0000 [r48249] Jason Parker <jparker@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48248 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48248 | qwell | 2006-12-04 15:48:41 -0600 (Mon, 04 Dec 2006) | 2 - lines Fix an issue which didn't allow unavail/greet/busy/etc - messages from being saved into ODBC (and probably IMAP). ........ - -2006-12-04 17:55 +0000 [r48229-48231] Jason Parker <jparker@digium.com> - - * /, configs/voicemail.conf.sample: Merged revisions 48230 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4 - lines Add documentation to voicemail.conf.sample for ODBC - storage. Issue 8499 - patch by blitzrage. ........ - - * /, doc/snmp.txt: Merged revisions 48228 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48228 | qwell | 2006-12-04 11:43:24 -0600 (Mon, 04 Dec 2006) | 4 - lines Attempt to document some of the dependencies that are - needed for net-snmp Issue 8499 - initial patch by blitzrage. - ........ - -2006-12-03 06:35 +0000 [r48224] Russell Bryant <russell@digium.com> - - * /, sounds/Makefile: Merged revisions 48223 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48223 | russell | 2006-12-03 01:34:14 -0500 (Sun, 03 Dec 2006) | - 3 lines When "fetch" is in use, instead of "wget", --continue is - not a valid option. (issue #8451) ........ - -2006-12-02 22:03 +0000 [r48200-48220] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Cleaning up handle_response a bit. - (Imported from 1.4) - - * .cleancount: Removing two .h files means we need to update - cleancount to force make depend again (or ?) - - * channels/chan_sip.c: Send CANCEL to call with early media - (PROGRESS INBAND). This is imported from branch "invitestate" and - "invitestate-1.4" *** *** *** IF YOU HAVE ISSUES WITH - BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN! *** Thank you! *** - *** /Olle - - * channels/chan_sip.c: Invitestate updates - - * agi/Makefile: Oops. Something is wrong in the agi directory. - Asking for autoconfig.h. I have it disabled locally, but no - reason to commit that change. - - * apps/app_sms.c: Doxygenification - - * main/coef_out.h (removed), main/tdd.c, main/callerid.c, - main/fskmodem.c, main/coef_in.h (removed): - Code formatting - - remove coef_in.h and coef_out.h that was only included as data - definitions in fskmodem.c If you understand spanish, please help - us translate the comments in fskmodem.c. Thanks! - - * /, channels/chan_sip.c, include/asterisk/rtp.h, - configs/sip.conf.sample, main/rtp.c: - Disable RTP timeouts - during T.38 transmission - Encapsulate RTP timers to the RTP - structure, so we have one set for video and one for audio - - Document RTP keepalive configuration option - Cleanup and - document the monitor support function to hangup on RTP timeouts - - Add RTP keepalive to SIP show settings Imported from 1.4 with - modifications for trunk. - -2006-12-01 23:39 +0000 [r48194] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c, /: Merged revisions 48193 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48193 | kpfleming | 2006-12-01 17:37:28 -0600 - (Fri, 01 Dec 2006) | 10 lines Merged revisions 48192 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 - Dec 2006) | 2 lines if Dial() is going to send music-on-hold to - the calling party, it has to send PROGRESS first to ensure that - the reverse audio path has been setup first (BE-106) ........ - ................ - -2006-12-01 23:20 +0000 [r48191] Russell Bryant <russell@digium.com> - - * Makefile, /, configure, configure.ac, makeopts.in, - sounds/Makefile: Merged revisions 48190 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) | - 12 lines FreeBSD 6.1 does not include wget by default. However, - it has fetch which will work just fine for our purposes of - downloading the sounds packages. So, check for both wget and - fetch and the configure script and use what was found to download - them. If neither one was found, and sound packages are selected - that must be downloaded, the install process will print out an - informative error message indicating the situation. Also, fix a - couple places where "make" was hard coded into some output - messages by replacing them with the $(MAKE) variable. (issue - #8451, initial patch by pabelanger, with additional modifications - by me) ........ - -2006-12-01 20:49 +0000 [r48188] Olle Johansson <oej@edvina.net> - - * main/channel.c: Formatting fix - -2006-12-01 20:26 +0000 [r48187] Jason Parker <jparker@digium.com> - - * /, configs/extensions.conf.sample: Merged revisions 48186 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, - 01 Dec 2006) | 10 lines Merged revisions 48183 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 - lines Fix a small typo - issue 8848, reported by pabelanger - ........ ................ - -2006-12-01 19:41 +0000 [r48180] Tilghman Lesher <tlesher@digium.com> - - * /, main/cli.c: Merged revisions 48179 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006) - | 2 lines Double-unlock error (reported by blitzrage on IRC) - ........ - -2006-12-01 18:16 +0000 [r48175-48178] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, configs/sip.conf.sample: - Remove T.38 - early media, since T.38 requires two way communication (imported - from 1.4) - Small fixes to limitonpeer - - * include/asterisk/threadstorage.h: Tiny doxygen improvement - -2006-11-30 21:22 +0000 [r48169] Joshua Colp <jcolp@digium.com> - - * /, include/asterisk/rtp.h, channels/chan_gtalk.c, main/rtp.c: - Merged revisions 48168 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 - lines Do not do a partial bridge for Google Talk since we need to - handle STUN. (issue #8448 reported by phsultan) ........ - -2006-11-30 20:55 +0000 [r48164-48167] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue #8319 (imported from 1.2, 1.4) - - Increment nonce-count properly (noriyuki) - - * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c, - include/asterisk/channel.h, include/asterisk/pbx.h: Documentation - updates - -2006-11-30 20:29 +0000 [r48153-48163] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 48158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, - 30 Nov 2006) | 10 lines Merged revisions 48157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 - lines Only print out debug message if bridged channel is not - NULL. (issue #8412 reported by jubilex) ........ ................ - - * /, res/res_features.c: Merged revisions 48155 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48155 | file | 2006-11-30 14:05:14 -0500 (Thu, - 30 Nov 2006) | 10 lines Merged revisions 48154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 - lines Do not listen for DTMF on the bridge that comes into - existence when ParkedCall is executed. This means native bridging - can now occur for this. (issue #8406 reported by kebl0155) - ........ ................ - - * main/cdr.c, /: Merged revisions 48152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48152 | file | 2006-11-30 13:47:40 -0500 (Thu, - 30 Nov 2006) | 10 lines Merged revisions 48151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 - lines Print certain CDR messages out at the NOTICE level versus - WARNING since they can occur when used with the CDR applications - and are perfectly fine. (issue #8367 reported by dartvader) - ........ ................ - -2006-11-30 18:25 +0000 [r48149-48150] Olle Johansson <oej@edvina.net> - - * main/devicestate.c: Small update - - * agi/Makefile, contrib/asterisk-ng-doxygen, agi/eagi-test.c, - main/devicestate.c, agi/eagi-sphinx-test.c: Doxygen updates - -2006-11-30 18:20 +0000 [r48144-48148] Joshua Colp <jcolp@digium.com> - - * /, configs/sip.conf.sample: Merged revisions 48143 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, - 30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 - lines Document 'port' for SIP peers, came up because of the - current mailing list thread. (issue #8450 reported by blitzrage) - ........ ................ - -2006-11-30 17:15 +0000 [r48130-48139] Olle Johansson <oej@edvina.net> - - * include/asterisk/doxyref.h, main/devicestate.c: Adding some - generic docs on extension and device states - watchers and - providers - - * doc/manager.txt, /: Add information on status events - - * /, channels/chan_sip.c: Merging patch from 1.2/1.4. I think this - was originally spotted by Luigi, but hit me in the back today. - -2006-11-30 03:29 +0000 [r48116-48123] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: I am pretty sure that oej only meant to - change the variable name in the source, not the configuration - option name so let's turn it back to srvlookup instead of - global_srvlookup. (issue #8442 reported by jtodd) - - * /, apps/app_voicemail.c: Merged revisions 48115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48115 | file | 2006-11-29 16:05:17 -0500 (Wed, 29 Nov 2006) | 2 - lines Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 - reported by slimey) ........ - -2006-11-29 20:57 +0000 [r48111-48114] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Clarify some settings for status - reports in subscriptions, queues and manager - - * /, configs/sip.conf.sample: Explain RTP timeouts - - * main/rtp.c: Change logging for p2p rtp bridge mode - -2006-11-29 17:59 +0000 [r48109-48110] Russell Bryant <russell@digium.com> - - * include/asterisk/threadstorage.h: - Fix a few spelling mistakes. - - Add some more documentation for the - ast_dynamic_str_............() function to document the behavior - of the function in the case of a partial write. Also, document - the return value and note that the function should never need to - be called directly. - - * main/utils.c: Go ahead and make this write unconditional. Making - it conditional is more work in both the append and non-append - modes. Also, always truncating the partial write makes the - behavior of the function more consistent, where in any type of - write, no partial result is left in the buffer. Thanks for the - feedback, luigi - -2006-11-29 16:53 +0000 [r48108] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 48107 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, - 29 Nov 2006) | 10 lines Merged revisions 48106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 - lines If the frame was duplicated before writing out then we need - to free it. (issue #8429 reported by edguy3) ........ - ................ - -2006-11-29 05:08 +0000 [r48103] Russell Bryant <russell@digium.com> - - * main/utils.c: Remove an XXX command suggesting that this - truncation should not be conditional, and also add a more verbose - comment explaining why it is only needed in the case of appending - to the string for any curious readers that come along in the - future. - -2006-11-29 04:28 +0000 [r48100-48102] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48101 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48101 | file | 2006-11-28 23:26:53 -0500 (Tue, 28 Nov 2006) | 2 - lines Don't crash if the mailstream was not created. ........ - - * sounds/Makefile: Use the proper version of extra sounds. (issue - #8441 reported by jtodd) - -2006-11-28 23:13 +0000 [r48098-48099] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Add a comment to note near some code that - performs a very expensive operation that occurs for every - incoming media frame. - - * codecs/codec_zap.c: resolve a couple of compiler warnings - -2006-11-28 18:28 +0000 [r48096] Jason Parker <jparker@digium.com> - - * Makefile, /: Merged revisions 48095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48095 | qwell | 2006-11-28 12:26:53 -0600 (Tue, 28 Nov 2006) | 2 - lines Export several more variables in top level Makefile. - Inspired by issue 8438. ........ - -2006-11-28 17:08 +0000 [r48090] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: don't use outputstr in the struct mansession, - it's just an extra allocation on a path where we have way too - many already. Unfortunately the AMI-over-HTTP requires multiple - copies, because we need to generate a header, then the raw output - to an intermediate buffer, then convert it to html/xml, and - finally copy everything into a malloc'ed buffer because that's - what the generic_http_callback interface expects. - -2006-11-28 16:59 +0000 [r48089] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 48088 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, - 28 Nov 2006) | 10 lines Merged revisions 48087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 - lines According to the research I have done we never needed to - include compiler.h in the first place so let's not! (issue #8430 - reported by edguy3) ........ ................ - -2006-11-28 15:53 +0000 [r48062-48086] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: initialize the dynamic string in a sane way. - - * main/utils.c: some simplifications to - ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name() - I am unsure whether the truncation of the string in case of a - failed attempt should be done unconditionally. See the XXX mark. - Russel, ideas ? - - * main/manager.c: do not return 500 Internal error if the AMI - command provides no output. - - * main/manager.c: mosty comment and documentation cleanup on - waitevent. - - * main/manager.c: Move the code to purge stale sessions to a - function, to simplify the body of the main loop of the accepting - thread. Rename purge_unused() to purge_events() so one knows what - the function does. - - * main/manager.c: Various simplifications of the code: + use a - wrapper around ast_carefulwrite(), used in two places, to make - life easier when we decide to use a different interface to the - socket. + put an ast_verbose() message on astman_append on a case - that should never happen now that we use a temporary file for - AMI-over-HTTP sessions + document and slightly simplify - process_events() by removing unnecessary parentheses. + in - get_input(), use ast_wait_for_input() instead of poll(). We may - want to move to a completely non-blocking - - * main/manager.c: More informative message on invalid commands. - - * main/manager.c: another normalization of AMI vs HTTP - identification. Should really define a macro IS_AMI(s) so it is - clear what we want to do. - - * main/manager.c: always use managerid to determine whether this is - an AMI or HTTP session, and document it. - - * main/http.c: In the previous commit i forgot to set the - poll_timeout to -1, causing the http threads to do busy waiting - around the socket... Fix the mistake, sorry for the - inconvenience! - - * main/http.c: document the support for running a server on TCP/TLS - and opening an SSL socket. We are almost ready to make this code - available to other modules. - - * main/http.c, configs/http.conf.sample: add a new http.conf - option, sslbindaddr. Because https is more secure than http, it - usually makes sense to keep this service more open than the one - on the unencrypted port. - - * main/http.c: in the helper thread, separate the FILE * creation - from the actual function doing work on the socket. This is - another generalization to provide a generic mechanism to open - TCP/TLS socket with a thread managing the accpet and children - threads managing the individual sessions. - - * main/http.c: staticize a global variable and remove an unused - field structure. - -2006-11-27 18:10 +0000 [r48056] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 48054 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48054 | file | 2006-11-27 13:06:50 -0500 (Mon, - 27 Nov 2006) | 10 lines Merged revisions 48053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 - lines Use the proper function to get the new message count - instead of always using the filesystem. (issue #8421 reported by - slimey) ........ ................ - -2006-11-27 17:31 +0000 [r48050] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 48049 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48049 | tilghman | 2006-11-27 11:20:37 -0600 - (Mon, 27 Nov 2006) | 10 lines Merged revisions 48045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 - Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) - ........ ................ - -2006-11-27 15:48 +0000 [r48039-48040] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_spool.c: More fixes for referencing a structure after it - has been freed. (issue #8425 reported by arkadia) - - * pbx/pbx_spool.c, /: Merged revisions 48038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r48038 | file | 2006-11-27 10:32:19 -0500 (Mon, - 27 Nov 2006) | 10 lines Merged revisions 48037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 - lines Do not reference the freed outgoing structure in the debug - message. (issue #8425 reported by arkadia) ........ - ................ - -2006-11-27 14:47 +0000 [r48034] Luigi Rizzo <rizzo@icir.org> - - * funcs/func_cdr.c: remove an extra comma in an initializer - Detected by: AST_DEVMODE=yes - -2006-11-27 06:59 +0000 [r48032-48033] Olle Johansson <oej@edvina.net> - - * include/asterisk/doxyref.h, include/asterisk/threadstorage.h: - Doxygen updates - - * /, channels/chan_sip.c: Change error message (imported from 1.4) - -2006-11-26 06:55 +0000 [r48019] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h, include/asterisk/threadstorage.h: - Add - some comments on thread storage with a brief explanation of what - it is as well as what the motivation is for using it. - Add a - comment by the declaration of ast_inet_ntoa() noting that this - function is not reentrant, and the result of a previous call to - the function is no longer valid after calling it again. - -2006-11-26 00:31 +0000 [r48016-48018] Steve Murphy <murf@digium.com> - - * /, funcs/func_cdr.c: Merged revisions 48017 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 - line might as well also document the raw values of the flag vars - ........ - - * /, funcs/func_cdr.c: Merged revisions 48015 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 - line A little bit of func_cdr documentation upgrade-- no bug# - involved, although 8221 may have inspired it. ........ - -2006-11-25 21:50 +0000 [r48009-48014] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Little fix so we use the right message - - * channels/chan_zap.c: Make compiler happier - - * channels/chan_zap.c: bug fix - - * channels/chan_zap.c: Make sure we don't send a group reset on a - group larger than 32 CICs - - * channels/chan_zap.c: Add ss7 show linkset command - - * channels/chan_zap.c: Updates to show linkset command - -2006-11-25 17:37 +0000 [r48008] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: generalize a bit the functions used to create an tcp - socket and then run a service on it. The code in manager.c does - essentially the same things, so we will be able to reuse the code - in here (probably moving it to netsock.c or another appropriate - library file). - -2006-11-25 09:48 +0000 [r48003-48004] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Changing ERROR to lesser level. Imported - from 1.2/1.4 - - * main/rtp.c: - Adding comment on suspicious memory allocation. - Seems like it's never freed, but I don't have a clear - understanding of the frame allocation/deallocation, so I just - mark this for investigation. (Reported by Ed Guy). We're trying - to see if a free() hurts... - Doxygen comments on p2p rtp bridge - stuff. I am a bit worried about shortcutting rtcp this way, but - will need feedback from rtcp gurus. This should work for video - calls too, and possibly UDPTL. - -2006-11-25 09:02 +0000 [r48001] Luigi Rizzo <rizzo@icir.org> - - * main/channel.c: set pointers to NULL after freeing memory to - avoid multiple free() probably 1.4/1.2 issue as well if someone - can look into that. - -2006-11-24 18:17 +0000 [r47995-47997] Steve Murphy <murf@digium.com> - - * /: removed the svnmerge-integrated property from trunk; it's - confusing svnmerge in newly created branches - - * /, main/translate.c: This fix inspired by a patch supplied in bug - 8189, which points out problems with the PLC code - -2006-11-24 14:00 +0000 [r47986] Olle Johansson <oej@edvina.net> - - * include/asterisk/doxyref.h, main/pbx.c, - include/asterisk/causes.h, include/asterisk/channel.h: Doxygen - update - Document cause codes - Document a bit more on channel - variables - global, predefined and local - Fix some doxygen in - channel.h. Adding one comment for two definitions does not work. - They won't be copied to each. - -2006-11-23 11:04 +0000 [r47957-47960] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Remove unused memory allocation - - * doc/asterisk-conf.txt: Document new configuration option. - -2006-11-22 21:49 +0000 [r47933-47945] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 47944 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2 - lines Video will never reach Packet2Packet bridging and can do - more harm then good. ........ - - * CHANGES: Clarify a bit more. - - * CHANGES: Need to update the CHANGES file as well for the maxfiles - option. - - * main/asterisk.c: Add support to set the maximum number of files - open when Asterisk loads using the 'maxfiles' configuration - option. (issue #7499 reported by rkarlsba) - -2006-11-22 11:28 +0000 [r47923] Olle Johansson <oej@edvina.net> - - * channels/chan_h323.c: Don't over-deprecate... :-) - -2006-11-22 05:49 +0000 [r47912] Mark Spencer <markster@digium.com> - - * main/manager.c: Restore some sense of security to manager - -2006-11-21 17:34 +0000 [r47898] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 47897 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2 - lines If we have the non standard G726-32 setting turned on we - want to return G726-32 to the SDP, not our AAL2 string. (issue - #8330 reported by voipgate) ........ - -2006-11-21 15:25 +0000 [r47893] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Treat 101 as 100, not 183 session - progress - -2006-11-21 11:53 +0000 [r47880-47881] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dial.c: better fix for the previous bug. In general this - code needs a deep revision, because the body of do_forward() - deletes/overwrites the output channel without freeing the resouce - in some cases, and without notifying the caller. Also, on FreeBSD - with MALLOC_OPTIONS set i am seeing various panics (duplicate - freee etc.) - - * apps/app_dial.c: do not ast_hangup() on a NULL channel. In the - original code this would happen in the case of o->forwards >= - AST_MAX_FORWARDS Likely an 1.2/1.4 isse as well - please someone - have a look, while I am hunting a few more similar panics now. - -2006-11-20 20:04 +0000 [r47866] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 47864-47865 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47864 | tilghman | 2006-11-20 14:01:58 -0600 (Mon, 20 Nov 2006) - | 2 lines Oops, merge missed release of odbc object ........ - ........ - -2006-11-20 19:52 +0000 [r47851-47861] Joshua Colp <jcolp@digium.com> - - * main/frame.c, /: Merged revisions 47860 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47860 | file | 2006-11-20 14:51:36 -0500 (Mon, - 20 Nov 2006) | 10 lines Merged revisions 47859 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 - lines Don't forget to byte swap if we are exiting the smoother - feed early. (issue #8287 reported by arturs) ........ - ................ - - * main/rtp.c: Use RTP/RTCP fds on the RTP structure, don't bother - storing them. - - * /, main/rtp.c: Merged revisions 47852 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2 - lines Only remove/destroy the RTCP I/O item if it exists. - ........ - - * apps/app_dial.c, /, apps/app_directed_pickup.c, - include/asterisk/channel.h, .cleancount: Merged revisions 47850 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2 - lines Use a separate variable in the channel structure to store - the context that the channel was dialed from. (issue #8382 - reported by jiddings) ........ - -2006-11-20 14:08 +0000 [r47847] Steve Murphy <murf@digium.com> - - * /: Erased the svnmerge-integrated prop from trunk. Please, in - your svnmerge-ings, don't let these props leak into the trunk or - branches. - -2006-11-20 11:46 +0000 [r47844-47846] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Update docs for videosupport - - * /, channels/chan_sip.c: Properly reset schedule items (rizzo) - -2006-11-19 04:22 +0000 [r47835-47836] Steve Murphy <murf@digium.com> - - * UPGRADE.txt: Added a few words to explain the change to AEL - concerning Gosub() - - * doc/ael.txt: Added a few words of explanation about macros - -2006-11-18 22:14 +0000 [r47822-47834] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: comments-only change: document a bit more when - manager events are delivered to the clients. - - * main/cdr.c, res/res_features.c, res/res_realtime.c: - ESS-ification. no need to bring this in 1.4, it is just code - cleanup - - * include/asterisk/cli.h, main/cli.c: Move this macro from cli.c to - cli.h so apps can use it without duplicating the macro or the - code: /*! * In many cases we need to print singular or plural * - words depending on a count. This macro helps us e.g. * printf("we - have %d object%s", n, ESS(n)); */ #define ESS(x) ((x) == 1 ? "" : - "s") - - * /, channels/chan_sip.c: Merged revisions 47823 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 - lines fix bug 7450 - Parsing fails if From header contains angle - brackets (the bug was only in a corner case where the < was right - after the opening quote, and the fix is trivial). ........ - - * channels/chan_oss.c: prevent the sound thread from consuming all - the available CPU doing busy-wait on the output audio device. As - it is set now, it tries to push a frame every 10ms, which is - still too frequent but avoids deep restructuring of the code - (which i should do, though). Note, this is only for ring tones, - regular audio coming from the network is still delivered as soon - as it is available. Eventually this could well end up in the 1.4 - branch, but since i am probably the only user of chan_oss there - isn't much urgency to do that. - -2006-11-17 23:18 +0000 [r47821] Steve Murphy <murf@digium.com> - - * include/asterisk/file.h, main/channel.c, res/res_features.c, - main/file.c, main/app.c, apps/app_directory.c, - apps/app_followme.c, apps/app_voicemail.c: This update fulfils - the request of bug 7109, which claimed the language arg to - ast_stream_and_wait() was redundant. Almost all calls just used - chan->language, and seeing how chan is the first argument, this - certainly seems redundant. A change of language could just as - easily be done by simply changing the channel language before - calling. - -2006-11-17 22:56 +0000 [r47815-47818] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: remove a debugging message - - * main/cli.c: convert "help" to new style, fix completion of - arguments past the first one that i broke earlier today. - - * main/cli.c: standardize "module show [like]" - -2006-11-17 21:51 +0000 [r47814] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample, apps/app_voicemail.c: Add ability - to notify an external application/script that the voicemail - password was, while also still changing the password - "internally". Issue 7371, initial patch by pdunkel, with - rewrite/config comments by me. Additional modifications (yay - bitmask) by pdunkel. - -2006-11-17 21:50 +0000 [r47813] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: describe a bit the patterns that you can have in the - commands, and add support for wildcard (spelled as '%'). On - passing fix a bug in the expansion code which was hidden and - appeared when implementing the wildcard The fix is just the line - 'src != argindex', in case someone wants to test this on 1.4 - - but i would just keep this in trunk. - -2006-11-17 20:46 +0000 [r47806] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Add ability to add custom queue log via manager - interface. Issue 7806, patch by alexrch, with slight - modifications by me. - -2006-11-17 18:26 +0000 [r47801] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add some sense of link state. Don't make - calls if link is down. Only reset if we're bringing the link up - for the first time. - -2006-11-17 12:26 +0000 [r47787-47790] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: merge the implemenmtation of "core set debug" and - "core set verbose". No externally visible changes. - - * channels/chan_oss.c: remove an unused function - - * channels/chan_oss.c: use the regexp cli support on some of the - command - - * include/asterisk/cli.h, main/cli.c: introduce a bit of regexp - support in the internal CLI api. Now you can specify a cli - command as "console autoanswer [on|off]" which means the on|off - argument is optional, or "console {mute|unmute}" which means the - mute|unmute argument is mandatory. The blocks in [] or {} do not - necessarily need to be at the end of the string. Completions for - the variant parts are generated automatically. This should - significantly simplify the implementation of the various - handlers. - -2006-11-17 01:05 +0000 [r47784] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we choose the last channel as the - dchannel if it's not E1 (for BRI). (#8077) Thanks Tzafrir. - -2006-11-16 23:20 +0000 [r47783] Jason Parker <jparker@digium.com> - - * apps/app_dial.c, /, apps/app_db.c: Merged revisions 47782 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2 - lines Fix a couple of typos. Initially pointed out by mrobinson. - ........ - -2006-11-16 23:05 +0000 [r47779] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: convert two entries to new style - -2006-11-16 23:00 +0000 [r47778] Kevin P. Fleming <kpfleming@digium.com> - - * /, doc/billing.txt: Merged revisions 47777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47777 | kpfleming | 2006-11-16 17:00:10 -0600 - (Thu, 16 Nov 2006) | 12 lines update documentation regarding IAX2 - transfers and CDRs Merged revisions 47776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) - | 2 lines update clearly wrong documentation regarding cdr_custom - ........ ................ - -2006-11-16 22:51 +0000 [r47775] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Remove the interim variable for range - modifications, and set it on the structure directly. Also move - the default checking to where it gets set initially. Fixes - suggested by file. - -2006-11-16 22:44 +0000 [r47772] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: convert some handlers to new style. - -2006-11-16 22:32 +0000 [r47771] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: Add ability to - modify range for dring matching. Issue #8369, patch by ssuehring, - modified slightly by me. - -2006-11-16 22:03 +0000 [r47769-47770] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: fix indentation - - * main/cli.c: remove an unused function - -2006-11-16 21:13 +0000 [r47763-47765] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 47764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2 - lines Compare technology using the pointers instead of a straight - comparison based on name. (issue #8228 reported by dean bath) - ........ - -2006-11-16 20:10 +0000 [r47759] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, configure.ac: Merged revisions 47758 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r47758 | kpfleming | 2006-11-16 14:09:10 -0600 (Thu, 16 - Nov 2006) | 2 lines check for pre-1.4 versions of Zaptel and - abort the configure script if found with an appropriate error - message ........ - -2006-11-16 19:29 +0000 [r47756] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Make it possible - to enable/disable onhold tracking, in order to make life easier - for realtime users. - -2006-11-16 18:32 +0000 [r47747-47752] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47751 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, - 16 Nov 2006) | 10 lines Merged revisions 47750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 - lines Because of the way chan_local is written we should be extra - careful and make sure our callback functions have a tech_pvt. - (issue #8275 reported by mflorell) ........ ................ - - * /, apps/app_meetme.c: Merged revisions 47748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47748 | file | 2006-11-16 12:52:48 -0500 (Thu, 16 Nov 2006) | 2 - lines Don't unreference the SLA object if there is no SLA object - in the devicestate callback. (issue #8354 reported by loloski) - ........ - - * /: Be gone 1.2 props! - -2006-11-16 17:15 +0000 [r47734-47746] Olle Johansson <oej@edvina.net> - - * /: Merging a fix that was already fixed. - - * channels/chan_sip.c: Merging implementation of invite states from - my "invitestate" branch for 1.2. This is a bit more clean - platform for the handling of BYE/CANCEL than what we had. It - might also need to changes in other parts of the code, since we - know the state of the INVITE transaction. Please observe that - this is is not dialog states at all, this is INVITE transaction - states. Hello Michael Proctor, and thank you! :-) - - * /: Block upgrade to UPGRADE - - * /, channels/chan_sip.c, configs/sip.conf.sample: - CANCEL never - uses authentication - Add docs on canreinvite - -2006-11-16 14:58 +0000 [r47727-47732] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: reduce indentation on a large function. - - * main/cli.c: use atomic instructions to update the inuse counters - for CLI entriesC. The lock is not protecting this field. I wonder - if the field should be declared 'volatile' as well. - - * main/cli.c: make kevin (and 64 bit machines) happy and remove a - cast from char* to int in handling the return values from - new-style handlers. On passing, note that - main/loader.c::ast_load_resource() always return 0 so errors are - not propagated up. I am not sure this is the intended behaviour. - -2006-11-16 08:18 +0000 [r47718] Paul Cadach <paul@odt.east.telecom.kz> - - * main/channel.c, /, funcs/func_channel.c, - include/asterisk/channel.h: Merged revisions 44809 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10 - Окт 2006) | 1 line CHANNEL() function sometime mix parameter and - value ........ - -2006-11-15 22:32 +0000 [r47713] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47712 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, - 15 Nov 2006) | 10 lines Merged revisions 47711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 - lines Make sure that the pvt structure exists before trying to do - fixup on Local channels. (issue #7937 reported by mada123, fix by - alamantia with mods by me) ........ ................ - -2006-11-15 21:57 +0000 [r47710] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 47709 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47709 | tilghman | 2006-11-15 15:56:55 -0600 (Wed, 15 Nov 2006) - | 2 lines Fix ODBC_STORAGE for when context is NULL ........ - -2006-11-15 21:36 +0000 [r47708] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 47707 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2 - lines We need to ensure timelimit stuff is included as well so - warnings get played. (issue #8050 reported by KNK) ........ - -2006-11-15 21:21 +0000 [r47706] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Hunting the initreq change for an ACK - -2006-11-15 20:59 +0000 [r47703-47704] TransNexus OSP Development <support@transnexus.com> - - * apps/app_osplookup.c: 1. Fix the bug that Asterisk hangs up the - calls if the OSP AuthRsp messages without destination protocol - infomation. 2. Fix the bug that Asterisk generats wrong dial - string (no in - IAX2/[username[:password]@]peer[:port][/exten[@context]][/options] - format) for IAX. 3. Add support for oh323 channel driver. 4. - Re-formate the code. - - * include/asterisk/astosp.h: 1. Re-format the code. - -2006-11-15 20:51 +0000 [r47702] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/file.c: Merged revisions 47701 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47701 | kpfleming | 2006-11-15 14:50:06 -0600 (Wed, 15 Nov 2006) - | 2 lines don't try to call fclose() if fopen() failed ........ - -2006-11-15 20:40 +0000 [r47700] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: - Don't reply to ACK - Improve SIP - history for debugging (Imported from 1.4) - -2006-11-15 20:28 +0000 [r47685-47694] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 47693 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47693 | kpfleming | 2006-11-15 14:27:38 -0600 - (Wed, 15 Nov 2006) | 12 lines Merged revisions 47677 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 - Nov 2006) | 4 lines ensure that message duration is included in - email notifications for forwarded messages (BE-96, fix by me - after corydon used his clue-bat on me) ensure that duration in - the message metadata is updated if prepending is done during - forwarding (related to BE-96) remove prototype for API call that - does not exist ........ ................ - - * /, main/config.c: Merged revisions 47690 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47690 | kpfleming | 2006-11-15 14:01:22 -0600 - (Wed, 15 Nov 2006) | 20 lines Merged revisions 47686,47688-47689 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006) - | 2 lines clear the category's variable tail pointer as well when - variables are detached from it ........ r47688 | kpfleming | - 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when - appending a list of variable to a category, ensure the tail - pointer points to the last variable in the list ........ r47689 | - kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2 - lines when re-writing the config file, don't repeat the path if - it hasn't changed ........ ................ - - * /, main/config.c: Merged revisions 47684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47684 | kpfleming | 2006-11-15 12:43:30 -0600 - (Wed, 15 Nov 2006) | 10 lines Merged revisions 47682 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 - Nov 2006) | 2 lines ouch... don't use printf, use - ast_log/ast_verbose ........ ................ - -2006-11-15 17:40 +0000 [r47662-47669] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_oss.c: fix indentation - - * main/cli.c: small simplifications and localization of variables. - - * main/cli.c: new-style "core show channels" - - * main/cli.c: more changes to new style of "module load" and - "load". Under FreeBSD, the filename_completion used in the above - commands does not work. Not sure why, but on passing i note that - the function is part of readline and is not reentrant, so it - needs to be fixed one way or another. - - * main/cli.c: move another deprecated command to the new style - - * main/cli.c: move "core set debug" to the new style, and remove - duplicated code for the deprecated handler. On passing fix a long - standing bug in find_cli() which would not find the longest match - - this part (trivial, basically just update matchlen on a match) - must go in 1.4 and possibly 1.2 as well. - -2006-11-15 16:09 +0000 [r47657-47661] Olle Johansson <oej@edvina.net> - - * /: Messed up earlier, cleaning up... - - * /, channels/chan_sip.c: Send proper SIP error message improperly - when we can't allocate dialog (out of file handles is one cause) - - * channels/chan_sip.c: Update doxygen docs to reflect the code... - -2006-11-15 15:02 +0000 [r47652-47654] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/cli.h, main/cli.c: one more step cleaning the - internal CLI interface: the NEW_CLI macro now supports extra - arguments (to deprecate other commands). use this to implement - unload and reload, and remove some unused functions. usual - completion fixes (as these function accept multiple arguments). - The summary is still a bit inconsistent. - - * include/asterisk/cli.h, main/cli.c: update the internal cli api - following comments from kevin. This change basically simplifies - the interface of the new-style handler removing almost all the - tricks used in the previous implementation to achieve backward - compatibility (which is still present and guaranteed.) - -2006-11-15 04:47 +0000 [r47646] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 47645 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2 - lines If NAT detection is turned on or already detected then say - NAT is active when setting the remote RTP peer when doing early - bridging. (issue #8365 reported by marcelbarbulescu) ........ - -2006-11-15 00:19 +0000 [r47642] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/term.c: Merged revisions 47641 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47641 | kpfleming | 2006-11-14 18:19:05 -0600 (Tue, 14 Nov 2006) - | 2 lines more formatting cleanup, and avoid running off the end - of the string ........ - -2006-11-15 00:15 +0000 [r47640] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 47639 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2 - lines Turn notice about unknown RTCP packet type into a debug - message instead. ........ - -2006-11-15 00:06 +0000 [r47636] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/misdn/isdn_lib.c: Merged revisions 47635 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14 - Nov 2006) | 2 lines silence compiler warning on 64-bit platforms - (this variable is an 'int' anyway, comparing it to 'signed long' - is not useful) ........ - -2006-11-14 22:19 +0000 [r47633] Joshua Colp <jcolp@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 47632 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47632 | file | 2006-11-14 17:17:16 -0500 (Tue, - 14 Nov 2006) | 10 lines Merged revisions 47631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 - lines Update copyright information in the ADSI logo blob. - ........ ................ - -2006-11-14 22:08 +0000 [r47630] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: add missing casts and remove an unused function. - -2006-11-14 22:07 +0000 [r47623-47629] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 47628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2 - lines Only keep the video RTP structure around if 1. Video - support is enabled and 2. A video codec is enabled on the dialog - ........ - - * /, funcs/func_uri.c: Merged revisions 47625 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47625 | file | 2006-11-14 16:30:44 -0500 (Tue, 14 Nov 2006) | 2 - lines Small documentation clarification for URIENCODE. (issue - #8294 reported by salaud) ........ - - * apps/app_dial.c: Make local copy of arguments to parse. (issue - #8362 reported by homesick) - -2006-11-14 18:58 +0000 [r47622] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 47621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47621 | tilghman | 2006-11-14 12:54:40 -0600 (Tue, 14 Nov 2006) - | 3 lines Conversion of res_odbc API to include ast_ prefix did - not completely transition app_voicemail when ODBC_STORAGE is used - (reported on IRC by caio1982, not in bugtracker) ........ - -2006-11-14 17:00 +0000 [r47619-47620] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: fix completion for "module reload mod_1 mod_2 ... " - (should do the same for the other similar commands, "reload", - "module unload" and so on. - - * main/cli.c: partly convert to new style set-verbose, with - completion fixes - -2006-11-14 16:48 +0000 [r47618] Joshua Colp <jcolp@digium.com> - - * /, apps/app_amd.c: Merged revisions 47617 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2 - lines Use LOG_DEBUG to print out the indication that app_amd is - using default settings instead of using LOG_NOTICE. This stops - needless logging of this information under normal circumstances. - (issue #8361 reported by Seb7) ........ - -2006-11-14 16:38 +0000 [r47614-47616] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: replace two deprecated functions with calls to the - standard ones, with fixes to argc/argv - - * main/cli.c: remove duplicated implementation for a deprecated - function, use the original one instead with appropriate changes - in argc/argv. This is not always applicable, but in some simple - cases it is. - -2006-11-14 16:15 +0000 [r47610-47611] Olle Johansson <oej@edvina.net> - - * include/asterisk/cli.h: need to check quoting in the doxygen - docs... - - * include/asterisk/cli.h: Some improvements to doxygen docs - -2006-11-14 16:09 +0000 [r47606-47609] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: new-style for 'core show uptime', include 'complete' - support and better error checking - - * main/cli.c: apply previous fix to old-style cli entries as well. - - * main/cli.c: fix "core set debug atleast ", and fix the simple - case where a command can have multiple completions, the first - ones coming from keywords in previous CLI entries. There is no - solution yet for the complex case of N1 completions from a CLI - entry, followed by N2 from the next one, and so on, because the - _complete() handlers do not tell us how many matches it - generates, so we don't know how many to skip when interrogating - the other handlers. The only solution is to avoid, as much as - possible, multiple CLI entries with the same prefix. - - * include/asterisk/cli.h, main/cli.c: Bring in the improved - internal API for the CLI. WATCH OUT: this changes the binary - interface (ABI) for modules, so e.g. users of g729 codecs need a - rebuilt module (but read below). The new way to write CLI - handlers is described in detail in cli.h, and there are a few - converted handlers in cli.c, look for NEW_CLI. After converting a - couple of commands i am convinced that it is reasonably - convenient to use, and it makes it easier to fix the pending CLI - issues. On passing, note a bug with the current 'complete' - architecture: if a command is a prefix of multiple CLI entries, - we miss some of the possible options. As an example, "core set - debug" can continue with "channel" from one CLI entry, and "off" - or "atleast" from another one. We address this problem in a - separate commit (when i have figured out a fix, that is). ABI - issues: I asked Kevin if it was ok to make this change and he - said yes. While it would have been possible to make the change - without breaking the module ABI, the code would have been more - convoluted. I am happy to restore the old ABI (while still being - able to use the "new style" handlers) if there is demand. - -2006-11-14 13:17 +0000 [r47595-47600] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Adding some debug output to trace bug in - realtime - - * channels/chan_sip.c: Adding a new debug line for issue #7351 - - trying to find where an ACK can overwrite the initreq... - - * /, channels/chan_sip.c: Issue #8272 imported from 1.2/1.4 - Let - the peerpoke system destroy it's own packets, please. - - * channels/chan_sip.c: Remove flags not used any more (thanks - Luigi) - -2006-11-13 22:40 +0000 [r47586-47587] Matt O'Gorman <mogorman@digium.com> - - * codecs/codec_zap.c: oops no parens - - * main/frame.c, codecs/codec_zap.c: fix bytesize to 5.3kb for g723 - codec and add support for multimode of tc400p - -2006-11-13 21:32 +0000 [r47585] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 47584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47584 | file | 2006-11-13 16:28:57 -0500 (Mon, - 13 Nov 2006) | 10 lines Merged revisions 47583 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 - lines Initialize global pointers for connection and result to - NULL. (issue #8356 reported by james) ........ ................ - -2006-11-13 20:21 +0000 [r47582] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 47581 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47581 | tilghman | 2006-11-13 14:20:01 -0600 - (Mon, 13 Nov 2006) | 10 lines Merged revisions 47580 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 - Nov 2006) | 2 lines Having more than 255 old messages caused - corruption in the new/old count ........ ................ - -2006-11-13 19:20 +0000 [r47579] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Small fix for uncommon scenario. - -2006-11-13 19:19 +0000 [r47577-47578] Steve Murphy <murf@digium.com> - - * /: Blocking 47576 from merging into trunk. Already done in 47577 - - * main/config.c: This solves bug 8342, whereby a crash occurs under - certain circumstances while reading a config file with comments-- - a call to CB_ADD shouldn't happen if withcomments is zero - -2006-11-13 19:14 +0000 [r47575] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Make chan_h323 build again and make the CLI - commands work. (reported on asterisk-dev mailing list by Di-Shi - Sun) - -2006-11-13 18:24 +0000 [r47568] Steve Murphy <murf@digium.com> - - * /: blocked 47564 from 1.4 to be merged onto trunk; 47566 already - did it - -2006-11-13 18:23 +0000 [r47567] Joshua Colp <jcolp@digium.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add - 'loose' option to joinempty and leavewhenempty which is almost - exactly like 'strict' except it does not count paused queue - members as unavailable. (issue #8263 reported by gnarf) - -2006-11-13 18:20 +0000 [r47566] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing - the messed if, but we all forgot to update the regressions. Until - now. - -2006-11-13 17:55 +0000 [r47556-47560] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Don't play the "entering conference number - <insert number here>" prompts if the 'q' option is used. If - others believe this should be in 1.2/1.4 then we can put it in, - but I'm uncomfortable doing so right now as it is a change of - behavior. (issue #8138 reported by tmancill) - - * pbx/pbx_ael.c: Clean up last commit to better conform to - standards. - -2006-11-13 17:36 +0000 [r47554-47555] Steve Murphy <murf@digium.com> - - * /: Blocking 47553 from 1.4 to trunk... 47554 is done for it. - - * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being - found... just confuses users - -2006-11-13 17:10 +0000 [r47543-47552] Joshua Colp <jcolp@digium.com> - - * /, apps/app_sms.c: Merged revisions 47551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47551 | file | 2006-11-13 12:08:07 -0500 (Mon, - 13 Nov 2006) | 10 lines Merged revisions 47549 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 - lines When sending an SMS with a user data header properly set - the UDH flag in the first byte. (issue #8347 reported by - hoffmeis) ........ ................ - - * main/cli.c: Return module show to a working state. (issue #8353 - reported by jserve) - -2006-11-13 16:08 +0000 [r47541] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Only produce error message once, don't - fill the screen with them... (Testing SIPP thanks to JerJer and - Greg) - -2006-11-13 14:29 +0000 [r47536-47539] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: merge from astobj2-r47450: use UNLINK to - remove a packet from its queue, and related code rearrangement. - Approved by: oej This could be made better if we declared struct - sip_pvt *dialpg = pkt->owner; at the beginning of the function, - and use it throughout the function. I'll let the boss decide :) - - * channels/chan_sip.c: merge from codename-pineapple and astobj2 - 47499: simplify __sip_ack() removing a strcmp for looking up - packets. no functional change, only performance, so don't need to - merging to earlier branches now. Approved By: oej - - * main/cli.c: stop looking for new entries when we know we are - done. there is no functional change, so it is not necessary to - bother merging this to 1.4 now. - - * main/cli.c: free memory when unregistering an entry. needs to be - merged to 1.4 - -2006-11-13 05:58 +0000 [r47530] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample: Feature: allow the - sanity SQL to be customized per connection class (Issue 6453) - -2006-11-13 05:51 +0000 [r47529] Russell Bryant <russell@digium.com> - - * /, configure, acinclude.m4: Merged revisions 47527 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r47527 | russell | 2006-11-13 00:48:18 -0500 (Mon, 13 - Nov 2006) | 5 lines AC_PROG_SED is included in autoconf 2.60, but - apparently it is not included in 2.59. So, to maintain - compatability with 2.59 since it is a small change, copy this - macro into acinclude.m4 and rename it to AST_PROG_SED. (issue - #8345) ........ - -2006-11-13 05:48 +0000 [r47524-47528] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 47526 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47526 | tilghman | 2006-11-12 23:46:18 -0600 - (Sun, 12 Nov 2006) | 10 lines Merged revisions 47525 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 - Nov 2006) | 2 lines If the execute fails a second time, make sure - that we don't pass back a stale handle ........ ................ - - * channels/chan_zap.c, /: Merged revisions 47523 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47523 | tilghman | 2006-11-12 19:12:01 -0600 - (Sun, 12 Nov 2006) | 10 lines Merged revisions 47522 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 - Nov 2006) | 2 lines Don't play dialtone if the seizing the - channel fails (Bug 7754) ........ ................ - -2006-11-12 20:47 +0000 [r47521] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Part of patch in #7403 to improve tag - checking in pedantic mode (stephen_dredge) - -2006-11-12 19:22 +0000 [r47520] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: The use of an ifdef to check for FreeBSD is - useless here because the two versions of the format string are - identical. Also, since each format is only used once, get rid of - the use of defines all together. (issue #8344, julieng) In - passing, also clean up the formatting a but to get rid of the - nesting without the use of braces, as defined in the coding - guidelines. - -2006-11-12 16:15 +0000 [r47508-47514] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Restore auto-framing (DEA). Imported from - 1.4 - - * /, channels/chan_sip.c: - Support UDPTL as well as udptl in T38 - sdp. - - * /, channels/chan_sip.c: - Don't hangup because of failed - re-invite. Go back to previous state. - Keep RTP running during - T.38 session We might improve the code to issue ast_rtp_stop if - T.38 re-invite not fails later on in the code, but I don't see - many reasons to. - - * /, channels/chan_sip.c: - Add some comments to t.38 code - Remove - improper blocking of ptime: in SDP - -2006-11-12 06:31 +0000 [r47493-47498] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 47497 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47497 | russell | 2006-11-12 01:23:23 -0500 - (Sun, 12 Nov 2006) | 12 lines Merged revisions 47496 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 - Nov 2006) | 4 lines Only do the check to determine whether the - channel calling this function is an IAX2 channel when getting the - IP address using the special argument, CURRENTCHANNEL. (issue - #8341, jcovert) ........ ................ - - * Makefile, /: Merged revisions 47494 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47494 | russell | 2006-11-11 10:31:08 -0500 (Sat, 11 Nov 2006) | - 6 lines Add the target "menuconfig" as an alias for the - "menuselect" target. This is just a favor to users so that if you - accidentally type "make menuconfig" instead of "make menuselect", - it still works. (inspired by a comment on IRC from wangster - calling me an "especially devious asterisk developer" for having - it be menuselect instead of menuconfig. :) ) ........ - - * /, main/term.c: Merged revisions 47492 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47492 | russell | 2006-11-11 10:18:02 -0500 (Sat, 11 Nov 2006) | - 2 lines Tweak the formatting of this new function to better - conform to coding guidelines. ........ - -2006-11-11 02:12 +0000 [r47491] Matt O'Gorman <mogorman@digium.com> - - * main/logger.c, include/asterisk/term.h, main/term.c: safe - terminal output is sweet. - -2006-11-10 22:06 +0000 [r47478] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we don't use 32bits for a value - that only requires 1 bit. Also, fix a compiler warning for one of - the SS7 functions. - -2006-11-10 21:55 +0000 [r47467-47477] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Add some history and fix some debug - output for autodestruct. - - * /, channels/chan_sip.c: Proper fix for adding debug... - - * /, channels/chan_sip.c: Make sure we destroy dialog in case of - loop - - * /, channels/chan_sip.c: Cleanup imported from 1.4 - -2006-11-10 20:05 +0000 [r47459-47465] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_dundi.c: Fine, take this. - - * main/cli.c: A trunk that builds is a productive trunk. - - * pbx/pbx_dundi.c: Hello compiler working, goodbye compiler - warning. (fix compiler warning introduced from pbx_dundi - optimizations) - - * /, channels/chan_h323.c: Merged revisions 47457 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47457 | file | 2006-11-10 14:36:25 -0500 (Fri, 10 Nov 2006) | 2 - lines Let's give this a go... ........ - -2006-11-10 19:35 +0000 [r47456] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add fix for 7321. Ability to hide - calleridname from zapata.conf - -2006-11-10 19:01 +0000 [r47455] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue 8336- fix support for multipart SDP - (imported from 1.2/1.4). (Alphaque) - -2006-11-10 17:22 +0000 [r47445] Luigi Rizzo <rizzo@icir.org> - - * build_tools/prep_moduledeps: manual merge from 1.4: grep -m not - available on bsd, use head -1 which works for all - -2006-11-10 17:01 +0000 [r47439] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_h323.c, channels/chan_iax2.c, channels/chan_mgcp.c, - main/cli.c: Merged revisions 47436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006) - | 2 lines Discussion of these CLI changes resulted in more - consistency (Bug 8236) ........ - -2006-11-10 16:55 +0000 [r47438] Joshua Colp <jcolp@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 47437 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47437 | file | 2006-11-10 11:53:16 -0500 (Fri, 10 Nov 2006) | 2 - lines Only split up extension and context if a value exists. - (issue #8332 reported by loloski) ........ - -2006-11-10 16:38 +0000 [r47434-47435] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_queue.c: Merged revisions 47433 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47433 | kpfleming | 2006-11-10 10:36:49 -0600 (Fri, 10 Nov 2006) - | 2 lines if adding a queue member is LOG_NOTICE, then removing - them should be LOG_NOTICE, not LOG_DEBUG ........ - - * /, apps/app_queue.c: Merged revisions 47432 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47432 | kpfleming | 2006-11-10 10:34:04 -0600 (Fri, 10 Nov 2006) - | 2 lines reflect addition/removal of dynamic queue members in - queue_log, so that people using dialplan replacement for - AgentCallbackLogin can still track login/logout (issue #7736, - reported/patched by whoiswes but this commit was written by me - and covers all three paths for AQM/RQM) ........ - -2006-11-10 13:14 +0000 [r47415-47419] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Ripping out bad support for 491 replies - to INVITE's. Let's try again properly later. - - * /, channels/chan_sip.c: Fix badly defined flag. Thanks fenlander! - - * channels/chan_sip.c: Small simplification and documentation - correction. - -2006-11-10 04:30 +0000 [r47408-47410] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Various little bits of code cleanup to reduce - nesting, remove useless casts, and to remove a duplicated error - message after a memory allocation error - - * include/asterisk/app.h, apps/app_read.c, main/app.c: Add the - ability to specify multiple prompts to the Read() dialplan - application, similar to Background() and Playback(). (issue - #7897, jsmith, with some modifications) - -2006-11-10 03:45 +0000 [r47399-47406] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_h323.c: Merged revisions 47405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47405 | file | 2006-11-09 22:44:36 -0500 (Thu, 09 Nov 2006) | 2 - lines Fix building of chan_h323 by completeing some structure - definitions. (issue #8327 reported by Mithraen) ........ - - * main/pbx.c: This should already be called while locked. - - * /, apps/app_voicemail.c: Merged revisions 47398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47398 | file | 2006-11-09 17:32:30 -0500 (Thu, 09 Nov 2006) | 2 - lines Do conversion in a more easier to read and working way for - \r, \n, and \t. (issue #8324 reported by johnlange) ........ - -2006-11-09 21:32 +0000 [r47392] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /, build_tools/prep_moduledeps, - apps/app_voicemail.c: Merged revisions 47391 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | - 7 lines Work around an issue that caused menuselect to display a - bogus description for app_voicemail and chan_zap. These modules - use some preprocessor directives to determine what it will report - to Asterisk as its description. However, the way we extract this - information from the source files for menuselect is not smart - enough to figure this out. (issue #8326, #8328) ........ - -2006-11-09 17:08 +0000 [r47382] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 47380 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, - 09 Nov 2006) | 10 lines Merged revisions 47379 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 - lines Don't include compiler.h on kernels 2.6.18 and higher as, - well, it's apparently going to be removed. This should make all - you FC6 fans happy as your Asterisk will now build without any - mods. ........ ................ - -2006-11-09 16:30 +0000 [r47353-47378] Russell Bryant <russell@digium.com> - - * /, main/cli.c: Merged revisions 47377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47377 | russell | 2006-11-09 11:28:15 -0500 (Thu, 09 Nov 2006) | - 2 lines fix tab completion for "core debug channel" and "core no - debug channel" ........ - - * /, main/cli.c: Merged revisions 47375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47375 | russell | 2006-11-09 11:24:02 -0500 (Thu, 09 Nov 2006) | - 3 lines Fix "core show channel". Also, fix tab completion for - both "core show channel" and "core show channels". ........ - - * /, main/cli.c: Merged revisions 47372 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47372 | russell | 2006-11-09 11:18:33 -0500 (Thu, 09 Nov 2006) | - 3 lines Fix "core debug channel <whatever>". I guess someone - needs to go through and audit every CLI command that changed - number of arguments ... ........ - - * /, main/cli.c: Merged revisions 47366 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47366 | russell | 2006-11-09 10:49:39 -0500 (Thu, 09 Nov 2006) | - 3 lines Fix another CLI command, "core show uptime" ... (issue - #8323, reported by johnlange, fixed by myself) ........ - - * /, main/asterisk.c: Merged revisions 47352 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47352 | russell | 2006-11-09 01:31:37 -0500 (Thu, 09 Nov 2006) | - 3 lines fix "core show version" to reflect the new number of - arguments for this CLI command (issue #8316, kshumard) ........ - -2006-11-09 00:46 +0000 [r47343-47351] Steve Murphy <murf@digium.com> - - * /: Blocking 47344 from automerging into trunk - - * /: Blocking 47348 from automerging into trunk - - * main/channel.c: This mod via bug 7531 - - * channels/chan_skinny.c: committed in behalf of bug 8190 - -2006-11-08 22:35 +0000 [r47341] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Add Max-Forwards higher in the packet, - following recommendations - Fix documentation for - sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf - !!! - Change doc for a sip_pvt setting - -2006-11-08 21:59 +0000 [r47337-47339] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c: restore display of G.722 codec - - * /, channels/chan_sip.c: Merged revisions 47333 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006) - | 2 lines add simple fix for SDP to report proper sample rate for - G.722 media sessions ........ - -2006-11-08 18:26 +0000 [r47335] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, CHANGES: Display CID matching information when using - dialplan show. (issue #8279 reported by caio1982) - -2006-11-08 17:06 +0000 [r47325-47332] Russell Bryant <russell@digium.com> - - * /, utils/streamplayer.c: Merged revisions 47331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47331 | russell | 2006-11-08 12:03:09 -0500 (Wed, 08 Nov 2006) | - 5 lines I occasionally get email from users that are trying to - figure out what this does, or due to some misunderstanding as to - what it is supposed to do, can't get it to work. So, I have added - some text here to hopefully explain what this application does - and does not do. ........ - - * /, configure, configure.ac, acinclude.m4: Merged revisions 47327 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47327 | russell | 2006-11-08 11:31:59 -0500 (Wed, 08 Nov 2006) | - 4 lines Copy the macros from libtool.m4 to our own acinclude.m4 - such that libtool is no longer required to be installed to be - able to generate the configure script. ........ - -2006-11-08 15:28 +0000 [r47321] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: coding guidelines, coding guidelines, coding - guidelines - -2006-11-08 13:59 +0000 [r47314-47318] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47271 - avoid doing p > 0 when p is a pointer; move a lock closer to the - place where it is needed Approved By: oej - - * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47246 Same - as for peers and users, replace ASTOBJ_UNREF(r, - sip_registry_destroy) with unref_registry(r); Approved By: oej - - * channels/chan_sip.c: merge from team/rizzo/astobj2, rev 47243, - 47244, 47245: Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with - unref_peer(peer); This places the name of the destructor in one - place only (where it should be), eliminates the chance of errors - in case you specify the wrong destructor, and also lets the - compiler do type checking on the argument, again helping with - keeping the code clean. Same for users. remove two duplicate - definitions. Approved By: oej - - * channels/chan_sip.c: merge rev.47224 from team/rizzo/astobj2: - hide dialoglist lock/unlocking in wrapper functions. Approved By: - oej - - * channels/chan_sip.c: silence compiler about uninitialized - variables. The compiler is wrong, but it has the last word. - -2006-11-08 08:01 +0000 [r47313] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) - -2006-11-08 07:21 +0000 [r47306] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_jingle.c, channels/chan_gtalk.c: fix compilation. - Overall i think the previous change to ast_channel_alloc() to - close bug 7506 should have been done by defining an - ast_set_callerid_noevent() function that does the setting without - generating the event. Lot less code duplication, and easier to - handle. - -2006-11-08 03:13 +0000 [r47304-47305] Russell Bryant <russell@digium.com> - - * configure.ac: add a comment about where AC_PROG_LD comes from - - * aclocal.m4 (removed), /: remove aclocal.m4 from the tree since it - is just an intermediate file created while generating the - configure script. - -2006-11-07 23:14 +0000 [r47295-47300] Luigi Rizzo <rizzo@icir.org> - - * main/asterisk.c: fix "core show profile" parsing. Needs to go in - 1.4 too, but ENOTIME now - - * apps/app_queue.c: %ld and time_t don't match, so cast the - argument to long to ease portability problems - -2006-11-07 21:47 +0000 [r47290] Steve Murphy <murf@digium.com> - - * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc, - channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, - channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, - channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, - main/channel.c, channels/chan_jingle.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_nbs.c, - include/asterisk/stringfields.h, channels/chan_mgcp.c, - apps/app_voicemail.c: A fair number of changes for the sake of - bug 7506 - -2006-11-07 20:16 +0000 [r47285-47288] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47287 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov - 2006) | 2 lines This is not the commit you are looking for... - ........ - - * channels/chan_local.c, /: Merged revisions 47284 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov - 2006) | 2 lines Make MOH work as it did before in chan_local, - without this then it can go funky when transfers and MOH are - involved. (issue #7671 reported by jmls) ........ - -2006-11-07 18:56 +0000 [r47280] Kevin P. Fleming <kpfleming@digium.com> - - * /, configs/musiconhold.conf.sample: Merged revisions 47279 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) - | 2 lines clean up sample config, and make native file playback - the more obvious default choice ........ - -2006-11-07 18:50 +0000 [r47278] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: rge overhaul to voicemail imap support. - Allows support for more imap servers, also a better - implementation of several parts of the original work. patch - provided by 8033 with major upgrades. minor differences from 1.4 - patch do to changes in app_voicemail - -2006-11-07 17:33 +0000 [r47269] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Break -> continue to make stuff work... - Thanks, Luigi! - -2006-11-07 14:25 +0000 [r47257-47259] Kevin P. Fleming <kpfleming@digium.com> - - * /: remove another broken property merge - - * /: remove properties that shouldn't be merged to this branch - - * /: use editable URL for menuselect, and switch to trunk - -2006-11-07 13:26 +0000 [r47251-47252] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: issue #8265 - don't reply to ACK. - Imported from 1.2, 1.4 - - * include/asterisk/frame.h: Stealing Tilghman's explanation from - the -dev list and producing documentation... - -2006-11-07 08:34 +0000 [r47242] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c: explain why ast_carefulwrite is written the way it - is, and also that it doesn't really work as claimed. - -2006-11-07 01:28 +0000 [r47232-47240] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 47239 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r47239 | russell | 2006-11-06 20:25:10 -0500 - (Mon, 06 Nov 2006) | 13 lines Merged revisions 47238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 - Nov 2006) | 5 lines If random order is enabled for files mode - music on hold, set a random initial position, instead of always - starting at the first file, and doing the random operation only - when switching to the next file. (bug reported by John Lange on - the asterisk-dev mailing list) ........ ................ - - * utils/check_expr.c: check for failure after call to calloc() - (issue #8295) - -2006-11-06 17:27 +0000 [r47230] Kevin P. Fleming <kpfleming@digium.com> - - * UPGRADE.txt: minor change to test live syncing - -2006-11-06 17:05 +0000 [r47229] Joshua Colp <jcolp@digium.com> - - * main/manager.c, utils/astman.c, include/asterisk/manager.h: Add - support for manager hooks, so you could fire off manager events - over IRC if you were crazy enough. (issue #5161 reported by anthm - with mods by moi) - -2006-11-05 01:04 +0000 [r47210-47213] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Make pbx_dundi compile again. Sorry. :( - - * configs/zapata.conf.sample: List ss7 with the rest of the valid - signalling types. Group SS7 options together and comment them out - by default. - -2006-11-04 22:16 +0000 [r47209] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't lock dialoglist if monitor runs - __sip_destroy. Hmmm. I did not change pbx_dundi and yet it - doesn't compile ;-) - -2006-11-04 22:08 +0000 [r47206-47207] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: use the AST_MODULE_LOAD_* return codes from - load_module() - - * pbx/pbx_dundi.c: simplify a couple of loops - -2006-11-04 21:48 +0000 [r47205] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move IP address selection for media out of - add_sdp - -2006-11-04 21:44 +0000 [r47204] Russell Bryant <russell@digium.com> - - * pbx/pbx_dundi.c: Do some minor cleanup to the section of code - that sets the EID by getting the mac address for an ethernet - interface - -2006-11-04 21:17 +0000 [r47200-47203] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Make srvlookup global_srvlookup to follow - the rest of the code - - * channels/chan_sip.c: Simplify previous patch - - * channels/chan_sip.c, configs/sip.conf.sample: Adding new config - option "limitpeersonly" to only apply call limits to the peer - side of a type=friend. This is for trying to support BJ in his - quest to solve some issues with the queue system and type=friend - objects. BJ: Please test! - - * /, channels/chan_sip.c: Importing patch for Invite/replaces from - 1.4 - -2006-11-04 18:12 +0000 [r47197-47198] Russell Bryant <russell@digium.com> - - * /, main/cli.c: Merged revisions 47196 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47196 | russell | 2006-11-04 13:10:22 -0500 (Sat, 04 Nov 2006) | - 2 lines Fix another bug in "core set debug" ... ........ - - * /, main/asterisk.c, main/cli.c: Merged revisions 47195 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47195 | russell | 2006-11-04 12:59:39 -0500 (Sat, 04 Nov 2006) | - 2 lines Really fix the "core set debug" and "core set verbose" - CLI commands. ........ - -2006-11-04 17:45 +0000 [r47194] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Reverting rev 47093 until we have an - agreement on how to implement this, if at all. - -2006-11-04 17:40 +0000 [r47193] Russell Bryant <russell@digium.com> - - * /, main/cli.c: Merged revisions 47192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47192 | russell | 2006-11-04 12:38:24 -0500 (Sat, 04 Nov 2006) | - 3 lines fix the "atleast" option to the "core set verbose" and - "core set debug" CLI commands ........ - -2006-11-04 11:00 +0000 [r47179-47189] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dial.c: move out another large block to a large - function, and document some possibly missing parts in the privacy - screening code. Now that it is more streamlined it is easier to - see differences in handling the various cases. Have not tested - the code in depth. - - * res/res_agi.c: useless cast removal... - - * main/logger.c: remove many unnecessary casts - - * main/app.c: remove a useless cast - - * configs/manager.conf.sample: document the "debug" parameter, and - the change manager list -> manager show - - * apps/app_dial.c: fix indentation of a block, and do minor - simplifications at the end of another one. - - * apps/app_dial.c: complete previous commit. - - * apps/app_dial.c: move another block into a function. On passing, - avoid two null-pointer string dereference while printing messages - (which are sometimes not fatal in some platforms, but still - wrong). These two lines at least should be merged to 1.4 once i - am done with all the changes here. - - * apps/app_dial.c: move a large block into a separate function. - Mark with XXX a possible bug in previous code which used the - wrong source in case of a forwarded call. the function - do_forward() needs to be split further, as the initial part is - replicated in another places (with some minor differences, most - likely forgotten when updating after the copy). - -2006-11-03 23:27 +0000 [r47178] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This fix introduced via bug 8233 - -2006-11-03 23:24 +0000 [r47160-47177] Luigi Rizzo <rizzo@icir.org> - - * apps/app_dial.c: another small set of simplifications - - * apps/app_dial.c: change HANDLE_CAUSE into a function. - - * apps/app_dial.c: remove redundant checks - - * apps/app_dial.c: start integrating the simplifications proposed - in bug 0005860, as usual a bit at a time to ease locating new - bugs or fixes worth merging into other branches. In this commit, - introduce a macro, S_REPLACE, that replaces a string possibly - freeing the previous value. In one of these places (see the - comment marked XXX) the previous code might leak memory - if so, - this ought to be merged in 1.4 The macro might be worth putting - in one of the global headers (e.g. include/asterisk/strings.h) as - the construct is used in a million places in the asterisk code. - -2006-11-03 19:15 +0000 [r47146] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: One has to create the path and filename in - order to copy a file there. (issue #8278 reported by davebath) - -2006-11-03 18:53 +0000 [r47072-47132] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c, include/asterisk/manager.h: add a new - cli/manager.conf option "debug" to enable/disable debugging code - in the manager. At the moment the debugging code is very - lightweight, if the option is enabled manager messages also carry - a sequence number and the info where they have been generated - e.g. SequenceNumber: 10 File: chan_sip.c Line: 11927 Func: - handle_response_register It is not worthwhile having this as a - compile time option right now, because the extra work involved at - runtime is just checking one variable. - - * channels/chan_zap.c: remove old/useless usecnt stuff - - * channels/chan_vpb.cc: remove old/useless usecnt stuff. I think - this module doesn't compile, anyways, because it has not been - updated to the new module interface. - - * main/cli.c: Fix "core show channels" and "core show modules". Not - sure it applies like this to 1.4 because of deprecate versions of - the same command(s). - - * res/res_jabber.c: move variable declarations to the beginning of - a block. - - * /: block other changes of mine already applied to trunk. - - * /: block more changes of mine already applied to trunk - - * /: blocking 47107 - - * /: blocking 47108 - - * channels/chan_sip.c: Save the 'From' header received in a - REGISTER message so we can show it e.g. in the Manager interface. - This information is available as a callerid (or something like - that) during a call, but not when a device is registered but - silent. It may be useful to have it available e.g. when - developing a user interface/operator panel, to map numbers to - names. experimental, so not committed to 1.4 - - * channels/chan_jingle.c, channels/chan_gtalk.c: remove useless - usecnt stuff - - * channels/chan_phone.c: remove useless usecnt stuff - - * channels/chan_alsa.c: remove useless usecnt stuff - - * channels/chan_agent.c: remove useless usecnt stuff - - * channels/chan_features.c: remove useless usecnt handling - - * channels/chan_skinny.c: remove useless usecnt handling code - -2006-11-02 23:55 +0000 [r47052-47054] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, /, channels/chan_skinny.c, res/res_agi.c, - channels/chan_h323.c, res/res_jabber.c, main/rtp.c: Merged - revisions 47053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) - | 2 lines More changes making the CLI more consistent with - "category verb arguments" (continuation of issue 8236) ........ - - * main/pbx.c, channels/chan_local.c, main/frame.c, - channels/chan_sip.c, /, res/res_features.c, res/res_crypto.c, - channels/chan_agent.c, res/res_musiconhold.c, apps/app_queue.c, - channels/chan_iax2.c, main/config.c, main/cli.c, main/channel.c, - main/manager.c, channels/chan_skinny.c, res/res_agi.c, - channels/chan_features.c, main/logger.c, main/file.c, - main/http.c, res/res_indications.c, main/image.c, res/res_odbc.c, - main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: - Merged revisions 47051 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) - | 2 lines Reverse change of "show" to "list" and make several - other commands more consistent with "category verb arguments" - ........ - -2006-11-02 21:40 +0000 [r47037] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, include/asterisk/pbx.h: Let's make - application/function/hint lists read/write lists... just for - kicks - -2006-11-02 21:34 +0000 [r47035] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Updates to do unblock correctly - -2006-11-02 20:24 +0000 [r46999-47021] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Move check for codec translators to an - earlier place in the call, so we can fail gracefully (imported - from 1.4) - - * /, channels/chan_sip.c: Disable code for not implemented - functionality (T38 over RTP/TCP) - -2006-11-02 18:34 +0000 [r46991-46994] Russell Bryant <russell@digium.com> - - * include/asterisk/astobj.h: Sure enough, some of the uses of - astobj are doing recursive locking. This doesn't work with - rwlocks, so, this is reverted for now. - - * include/asterisk/astobj.h: astobj was already set up to use read - and write locks. Now that we have wrappers for them, use them - here. - -2006-11-02 18:01 +0000 [r46967-46972] Joshua Colp <jcolp@digium.com> - - * main/translate.c: Convert translation core linked list over to - read/write based one, since it spends most of it's time only - reading. - - * include/asterisk/linkedlists.h: Add AST_RWLIST_* set of macros - which implement linked lists using read/write locks, the actual - list manipulation is still done via the old macros. - -2006-11-02 17:51 +0000 [r46966] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 46965 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46965 | russell | 2006-11-02 12:49:54 -0500 - (Thu, 02 Nov 2006) | 11 lines Merged revisions 46964 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 - Nov 2006) | 3 lines ignore files in a music on hold directory - that begin with '.' (issue #8249, cboie) ........ - ................ - -2006-11-02 16:51 +0000 [r46940] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: Set the AST_RWLOCK_INIT_VALUE to the - PTHREAD_RWLOCK_INIT_VALUE if it is available, that way outside - stuff can determine whether to use a constructor or deconstructor - for initialization instead of using the init value. - -2006-11-02 16:50 +0000 [r46939] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Changes to show blocked/unblocked states, as - well as in service, out of service state - -2006-11-02 16:45 +0000 [r46938] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 46937 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006) - | 2 lines don't send INVITE when we have determined that we can't - offer any audio formats due to lack of trancoding support (or - incorrect configuration) ........ - -2006-11-02 16:28 +0000 [r46931-46935] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/lock.h: I'm crazy so I will add this... pthread - rwlock wrappers, along with autoconf stuff that detects the - presence of the initializer and the ability to set the kind of - lock (in our case we rather like writer preferred locks so writer - starvation doesn't occur... but on something like Darwin we don't - get that) - - * /, channels/chan_sip.c: Merged revisions 46930 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46930 | file | 2006-11-02 11:06:39 -0500 (Thu, - 02 Nov 2006) | 10 lines Merged revisions 46920 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 - lines Repeat after me oej: I will at least make sure my code - compiles before I commit it. ........ ................ - -2006-11-02 16:03 +0000 [r46926] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add simple down event support - -2006-11-02 15:47 +0000 [r46906] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c: - find_free_chan_in_stack: cleanup buggy usage - -2006-11-02 15:31 +0000 [r46902] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't overwrite pkt->flags (imported from - 1.2/1.4) - -2006-11-02 14:15 +0000 [r46846-46886] Russell Bryant <russell@digium.com> - - * main/callerid.c: various whitespace changes to reduce indentation - and to better conform to formatting guidelines - - * main/callerid.c: Change the buffer used in callerid_feed() and - callerid_feed_jp() to be allocated on the stack using alloca() - instead of using malloc() since they are only used locally to - these functions. - - * /, main/say.c: Merged revisions 46857 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46857 | russell | 2006-11-01 18:01:48 -0500 (Wed, 01 Nov 2006) | - 2 lines fix saying one hundred and two hundred in hebrew (issue - #7810, eldadran) ........ - - * CHANGES: Add a couple of things to the CHANGES file - - * Makefile, /, configure, codecs/gsm/Makefile, configure.ac, - build_tools/strip_nonapi, makeopts.in: Merged revisions 46847 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46847 | russell | 2006-11-01 17:51:21 -0500 (Wed, 01 Nov 2006) | - 3 lines Fixes for cross-compilation on mips (issue #8058, - ywalther, with some modifications) ........ - - * aclocal.m4, /, build_tools/menuselect-deps.in, configure, - build_tools/embed_modules.xml, configure.ac: Merged revisions - 46845 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46845 | russell | 2006-11-01 17:32:12 -0500 (Wed, 01 Nov 2006) | - 5 lines Add a check in the configure script to determine whether - ld is GNU ld or not. This is needed because module embedding only - works for gnu ld. GNU ld is now listed as a dependency for all of - the module embedding options in menuselect. (issue #8143) - ........ - -2006-11-01 20:38 +0000 [r46823] Matt O'Gorman <mogorman@digium.com> - - * /, channels/chan_gtalk.c: Merged revisions 46822 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.4 ........ - r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) - | 2 lines bind address support from bug 8164 ........ - -2006-11-01 19:48 +0000 [r46801] Steve Murphy <murf@digium.com> - - * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to - accept longer strings or mass confusion and a lot of lost time is - the result - -2006-11-01 18:41 +0000 [r46782] Joshua Colp <jcolp@digium.com> - - * /, main/Makefile: Merged revisions 46780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2 - lines Force poll() emulation for Darwin to always be on. It's too - broken to consider being used. This resolves the console issue - OSX users have been seeing. I would have liked to autoconf this - but I haven't been able to come up with a test case that works. - Que sera. ........ - -2006-11-01 18:40 +0000 [r46779-46781] Russell Bryant <russell@digium.com> - - * doc/channelvariables.txt, pbx/pbx_dundi.c: Add the ability to - pass options to the Dial application when using the DUNDi switch - in the dialplan by setting the DUNDIDIALARGS channel variable. - (issue #8084, patch by bluecrow76, with small modifications and - documentation updates) - - * /, res/res_monitor.c: Merged revisions 46778 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46778 | russell | 2006-11-01 13:26:35 -0500 - (Wed, 01 Nov 2006) | 17 lines Merged revisions 46776 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 - Nov 2006) | 9 lines soxmix and Asterisk expect different file - extensions for certain formats. This was already handled for the - wav49 format. However, it was not handled for ulaw and alaw. I - fixed this in such a way that using the alternate extensions for - ulaw and alaw will only happen if we know we're calling soxmix, - and not a custom script defined using the MONITOR_EXEC variable. - The wav49 processing was left alone so that external scripts will - see no behavior change. (issue #7550, reported by mnicholson, - proposed patch by junky, committed fix is a bit different) - ........ ................ - -2006-11-01 18:26 +0000 [r46777] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 46775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 - lines It's another round of chan_iax2 fixes! Should hopefully fix - the deadlock issues people have been reporting. IAXtel now has - qualify turned on for 800 peers and it is handling it fine. - ........ - -2006-11-01 18:16 +0000 [r46759-46774] Steve Murphy <murf@digium.com> - - * CHANGES: OOps. forgot to add this to CHANGES - - * main/say.c, apps/app_voicemail.c: This introduces Brazilian - Portuguese via 7663 - - * main/config.c: Cleanups suggested by Russell. - -2006-11-01 17:09 +0000 [r46758] Luigi Rizzo <rizzo@icir.org> - - * res/res_features.c: move variable declaration in the middle of a - block - -2006-11-01 16:51 +0000 [r46745] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 46744 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46744 | russell | 2006-11-01 11:39:09 -0500 (Wed, 01 Nov 2006) | - 2 lines Prevent an infinite loop when config processing gets to a - jitterbuffer option ........ - -2006-11-01 00:07 +0000 [r46732] Matt O'Gorman <mogorman@digium.com> - - * res/res_features.c: change default return extension after parking - timeout. 6953 with minor changes. - -2006-10-31 22:19 +0000 [r46719] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/translate.c, include/asterisk/translate.h: Merged - revisions 46714 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46714 | kpfleming | 2006-10-31 15:47:48 -0600 (Tue, 31 Oct 2006) - | 2 lines add an API so that translators can activate/deactivate - themselves when needed ........ - -2006-10-31 22:07 +0000 [r46717-46718] Jason Parker <jparker@digium.com> - - * main/translate.c: Fix "core show translation" output. Issue - #8243, patch by Damin. - -2006-10-31 18:10 +0000 [r46683-46696] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_iax2.c: remove old/useless usecount handling - - * channels/chan_sip.c: remove old/useless usecount stuff. - - * channels/chan_oss.c: remove old/useless usecount management code. - -2006-10-31 15:22 +0000 [r46661] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix the new send text manager command. There is - no way this could have worked. - Check the channel name string - length to be zero, not non-zero - Check the message string length - to be zero, not non-zero - unlock the channel *after* calling - sendtext - -2006-10-31 13:56 +0000 [r46582-46650] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Set #define for TIMER T1 value - - * channels/chan_sip.c: Cleaning up code - - * funcs/func_enum.c, /, include/asterisk/enum.h, main/enum.c: Issue - #80898 - Restoring func_enum (otmar) - - * main/manager.c: Add manager sendtext action. (Issue 6131, ZX81 - - thanks!) - - * /, channels/chan_sip.c, configs/sip.conf.sample: Fix rport - handling. ...where did the 1.2 properties come from, really? - they're back. - - * /, channels/chan_sip.c: - If peer that register fails ACL, fail - him - Remove the 1.2 props I've set by mistake earlier - - * /: Block patch that only applies to 1.4 - - * main/loader.c: Take two, using find_resource on Kevin's - suggestion. Might need better locking support, giving up if we - can't get the lock. Right now, using existing locking in - find_resource - -2006-10-31 06:37 +0000 [r46556-46565] Russell Bryant <russell@digium.com> - - * apps/app_cdr.c: add author doxygen tag (issue #8241, kshumard) - - * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 46563 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46563 | russell | 2006-10-31 01:30:53 -0500 (Tue, 31 Oct 2006) | - 3 lines Start Asterisk later in the boot process to ensure it - starts after stuff like MySQL (issue #8253, Alric) ........ - - * /, main/utils.c: Merged revisions 46561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46561 | russell | 2006-10-31 01:19:56 -0500 - (Tue, 31 Oct 2006) | 11 lines Merged revisions 46560 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 - Oct 2006) | 3 lines When handling the case where the hostname is - just an IPV4 numeric address, be sure to set the address type. - (issue #8247, alexr) ........ ................ - - * /, res/res_agi.c: Merged revisions 46558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46558 | russell | 2006-10-31 01:14:13 -0500 - (Tue, 31 Oct 2006) | 11 lines Merged revisions 46557 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 - Oct 2006) | 3 lines fix some copy/paste bugs in the checking of - arguments for the "control stream file" AGI command (issue #8255, - mnicholson) ........ ................ - - * /, main/translate.c: Merged revisions 46554 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | - 5 lines Add a small tweak to the code that checks to see whether - destination formats are translatable based on the source format. - If we have already determined that there is no translation path - in one direction, don't bother checking the other direction. - ........ - -2006-10-30 23:11 +0000 [r46541] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, utils/astman.c: These changes submitted by moy - via bug 6992, to add a Dial 'End' event to asterisk. I include - some changes to astman to cover other events that have been - added. - -2006-10-30 22:27 +0000 [r46529] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/translate.c: Merged revisions 46526 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006) - | 3 lines when unregistering a translator, don't rebuild the - translation matrix unless needed when filtering formats out of an - offer, ensure we check for translation ability in both directions - ........ - -2006-10-30 21:56 +0000 [r46513-46514] Olle Johansson <oej@edvina.net> - - * funcs/func_module.c: show, list, view, display... whatever. - - * funcs/func_module.c (added), include/asterisk/module.h, - main/loader.c: Adding dialplan function IFMODULE, so you can - create dialplans that handle various PBX installations and checks - if a module is loaded before using it. example - IFMODULE(chan_sip3.so) issue #6671 in the bug tracker, finally - gone. Thanks to mithraen for keeping it updated. - -2006-10-30 21:46 +0000 [r46512] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 46511 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46511 | kpfleming | 2006-10-30 15:46:07 -0600 (Mon, 30 Oct 2006) - | 2 lines ensure that items removed from a list are always - unlinked from the list (next pointer set to NULL) ........ - -2006-10-30 21:22 +0000 [r46508-46509] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Update sip list to eventlist format. - - * main/pbx.c, main/manager.c, include/asterisk/manager.h: Issue - #3930 - Add manager command for listing dialplan (coded april - 2005, in bugtracker since) - -2006-10-30 21:11 +0000 [r46507] Joshua Colp <jcolp@digium.com> - - * /, configure, configure.ac: Merged revisions 46506 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r46506 | file | 2006-10-30 16:09:13 -0500 (Mon, 30 Oct - 2006) | 2 lines Don't explicitly link in crypt as it is not used - on some platforms. ........ - -2006-10-30 19:56 +0000 [r46476-46489] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Change name of - "contact" setting to "callback" which better reflects what it is - to the person that configures asterisk. That we use it internally - in the contact header is a totally different story. Still not - convinced this is a good option. - - * channels/chan_sip.c: Globals need the "global_" prefix in - chan_sip, and need to be reset to default value at reload. - -2006-10-30 18:17 +0000 [r46475] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 46474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46474 | file | 2006-10-30 13:13:07 -0500 (Mon, 30 Oct 2006) | 2 - lines We need to lock the pvt structure during retransmission as - another worker thread may be doing something as well. ........ - -2006-10-30 18:04 +0000 [r46466] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we give the linkset number, not - the offset in the linksets array - -2006-10-30 18:02 +0000 [r46461] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Small conversion to ast_channel_unlock - -2006-10-30 17:32 +0000 [r46459] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Specify which linkset we're getting the - messages from in the message - -2006-10-30 16:59 +0000 [r46439] Olle Johansson <oej@edvina.net> - - * main/rtp.c: In debug mode, recognize that someone is sending - zrtp, even though we can't do anything with it yet. Ideally a - first step would be a passthrough mode. - -2006-10-30 16:50 +0000 [r46436] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Don't make errors when we don't need them - -2006-10-30 16:33 +0000 [r46379-46434] Olle Johansson <oej@edvina.net> - - * include/asterisk/file.h, include/asterisk/doxyref.h, /, - channels/chan_sip.c, main/ast_expr2f.c, - include/asterisk/module.h, formats/format_ogg_vorbis.c, - main/app.c, include/asterisk/channel.h, include/asterisk/lock.h, - include/asterisk/frame.h, main/asterisk.c, apps/app_voicemail.c: - Issue 8246 Doxygen updates (kshumard) THANK YOU! - - * /: The RTCP patch started in trunk, so don't start all over again - :-) - - * main/asterisk.c: Small formatting changes - - * main/rtp.c: Bind RTCP to the same IP as RTP. I currently don't - see this as a bug that needs to be fixed in 1.4/1.2 too, but feel - free to backport if you see it that way. RTCP now binds to ALL IP - addresses on the host, RTP to a specific address. - - * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 - redirects. - - * /, channels/chan_sip.c: Issue #7608 - Notifications sent with - wrong content-type (imported from 1.2, 1.4) - - * /: Block patch from other branch - - * channels/chan_sip.c: Issues related to issue #7828 - segfault - with MWI subscriptions and realtime. - - * /, channels/chan_sip.c: - Fix the OUTGOING stuff (merge from 1.4) - - Make sure we UNREF authpeer when not needed - - * apps/app_voicemail.c: Spelling fix. - - * channels/chan_sip.c: Documentation update again - - * channels/chan_sip.c: Documentation update (I guess) - - * channels/chan_sip.c: Documentation correction - - * channels/chan_sip.c: maxtime is not needed any more now that we - actually set the T1 timer based on the qualify result. - - * /, channels/chan_sip.c: Only accept message once - - * channels/chan_sip.c: Adding documentation inspired by a virtual - drink with an anonymous man in New Jersey - - * channels/chan_sip.c: Don't duplicate function if not needed... - - removing transmit_reinvite_with_t38_sdp in favour of adding an - argument to transmit_reinvite_with_sdp - - * /, channels/chan_sip.c: Merge from 1.4 : Don't send 183 - reliably... - - * channels/chan_sip.c: - Don't lock the dialoglist during the whole - destruction of a single SIP dialog. Only lock when needed - when - we remove the dialog from the dialog list If this doesn't lead to - severe problems, it might help with some locking issues in - 1.4/1.2. - Remove the term "interface" as a synonym for a SIP - dialog. Sorry, Mark, but no one understands it... ;-) - -2006-10-28 16:39 +0000 [r46378] Joshua Colp <jcolp@digium.com> - - * utils/Makefile, /: Merged revisions 46377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46377 | file | 2006-10-28 12:37:44 -0400 (Sat, 28 Oct 2006) | 2 - lines Don't build muted on OpenBSD, it is not supported. ........ - -2006-10-27 19:28 +0000 [r46372] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Let's make sure we hold the mutex lock before - we go looking at values in the queue structure that could - potentially be changing while we're running. - -2006-10-27 19:04 +0000 [r46371] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 46370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46370 | russell | 2006-10-27 14:03:32 -0500 (Fri, 27 Oct 2006) | - 4 lines move the copy of the default settings to the global - settings back out of process_zap, so that they aren't overwritten - when process_zap is called multiple times ........ - -2006-10-27 18:59 +0000 [r46369] BJ Weschke <bweschke@btwtech.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: * Added - option to run macro when a queue member is connected to a caller, - see queues.conf.sample for details. * Added QUEUE_VARIABLES - function to set queue variables added setqueuevar and - setqueueentryvar options for each queue, see queues.conf.sample - for details. (#8216, jmls reported and submitted) - -2006-10-27 18:31 +0000 [r46368] Olle Johansson <oej@edvina.net> - - * /, contrib/asterisk-ng-doxygen: raise the pressure on Christian - :-) - -2006-10-27 17:46 +0000 [r46366] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: First pass at implementation to be able to - block and unblock zap channels for use. - -2006-10-27 17:45 +0000 [r46365] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Put this patch on hold pending further - testing... - -2006-10-27 17:42 +0000 [r46359-46364] Russell Bryant <russell@digium.com> - - * /, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c, - main/asterisk.c: Merged revisions 46363 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | - 5 lines We should always be using _exit() after a fork() or - vfork() instead of exit(). This is because exit() does some extra - cleanup which in some implementations of vfork(), for example, - can actually modify the state of the parent process, causing very - weird bugs or crashes. (issue #7971, Nick Gavrikov) ........ - - * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add - the ability to customize some of the prompts used within the - voicemail application by configuring them in voicemail.conf - (issue #7415, patch by fkasumovic, with some fixes and - documentation updates by myself) - - * channels/chan_zap.c, /: Merged revisions 46358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | - 5 lines Instead of iterating all of the options once to look for - jitterbuffer options, and then again for everything else, move - the processing of jitterbuffer options into the main loop so that - there are no erroneous messages about ignoring unknown options. - (issue #8226) ........ - -2006-10-27 11:18 +0000 [r46354] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, - channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample, - channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged - revisions 46351-46353 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r46351 | crichter | 2006-10-27 11:49:20 +0200 - (Fr, 27 Okt 2006) | 9 lines Merged revisions 46176 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 - Okt 2006) | 1 line added nttimeout option to configure wether we - disconnect calls on NT timeouts or not during an overlapdial - session ........ ................ r46352 | crichter | 2006-10-27 - 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line fixed not compile - issue, which was just introduced ................ r46353 | - crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines - Merged revisions 46350 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | - 1 line fixed a bug which caused chan_misdn to try to allocate 2 - times the same channel on high load, which then caused - instability of mISDN. removed a useless function from isdn_lib.c - ........ ................ - -2006-10-26 20:27 +0000 [r46348] Jason Parker <jparker@digium.com> - - * /, apps/app_page.c: Merged revisions 46347 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2 - lines Fix small formatting issue, that causes misaligned line - ........ - -2006-10-26 20:22 +0000 [r46346] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Show if the channel is ready for video or - T.38 udptl - -2006-10-26 18:04 +0000 [r46341] Jason Parker <jparker@digium.com> - - * contrib/scripts/astgenkey.8: oops - somebody forgot to change - this - long ago, probably. - -2006-10-26 17:52 +0000 [r46330-46339] Russell Bryant <russell@digium.com> - - * main/pbx.c, apps/app_osplookup.c, main/manager.c, - apps/app_meetme.c, apps/app_festival.c, main/say.c, - apps/app_alarmreceiver.c, apps/app_sms.c, apps/app_rpt.c, - main/rtp.c, apps/app_voicemail.c: fix various spelling mistakes - in comments (issue #8237, jmls) - - * /, main/translate.c: Merged revisions 46329 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | - 11 lines - If the source has no audio or no video portion, do not - call powerof() to get the format index. - Don't run through the - audio and video loops if there is no audio or video portion of - the source If 0 is passed to powerof, it will return -1. This - value of -1 was then being used as an array index in these loops, - which caused a crash on some systems. Other than this issue, this - code works as we expected it to. If a format is not in the - source, and we have to translation path to it, it is not offered - in the list of acceptable destination formats. (fixes issue - #8231) ........ - -2006-10-26 12:47 +0000 [r46308-46319] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: fix a problem that i recently introduced when the - manager receives long commands. - - * configs/sip.conf.sample: document the match_auth_username option - -2006-10-26 04:19 +0000 [r46299] Russell Bryant <russell@digium.com> - - * /, doc/backtrace.txt: Merged revisions 46298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46298 | russell | 2006-10-25 23:18:00 -0500 (Wed, 25 Oct 2006) | - 2 lines update backtrace documentation to reflect changes in 1.4 - (issue #8230, kshumard) ........ - -2006-10-26 01:38 +0000 [r46288] Mark Spencer <markster@digium.com> - - * main/manager.c, main/config.c: Fix comment preservation code - (thanks murf!) - -2006-10-25 20:21 +0000 [r46259-46277] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Old todo: Don't add Contact headers on - BYE and CANCEL. - - * channels/chan_sip.c: First stab at transaction direction fix, - this for trunk for testing - - * /, channels/chan_sip.c: Ugly code to try to remove issue - discovered by Luigi as well as attack bug #7608 - -2006-10-25 19:24 +0000 [r46256] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Send CPG when we get a CONTROL_PROGRESS - frame and make sure that it sends ACM (not CPG) when we get - CONTROL_PROCEEDING. - - -2006-10-25 19:14 +0000 [r46251] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, configs/zapata.conf.sample: Update changes - to do US style point code parsing/formatting (xxx.xxx.xxx) - -2006-10-25 19:10 +0000 [r46250] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 46249 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46249 | russell | 2006-10-25 14:08:18 -0500 (Wed, 25 Oct 2006) | - 2 lines update warning message to include "agi" option (issue - #8225, jmls) ........ - -2006-10-25 17:12 +0000 [r46238] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 46237 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46237 | kpfleming | 2006-10-25 12:08:58 -0500 (Wed, 25 Oct 2006) - | 2 lines add support for prebuilt G.722 prompts and music on - hold files ........ - -2006-10-25 16:01 +0000 [r46215-46224] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merge from 1.4 - - * /: Block change to 1.4 to block change to 1.2... This is - confusing, but I think I got it right. - -2006-10-25 14:55 +0000 [r46201-46203] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c, main/translate.c, - include/asterisk/translate.h: Merged revisions - 46082-46083,46152-46153 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) - | 2 lines add an API call to allow channel drivers to determine - which media formats are compatible (passthrough or transcode) - with the format an existing channel is already using ........ - r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) - | 2 lines ensure that the translation matrix is properly - lock-protected every place it is used ........ r46152 | kpfleming - | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines if - multiple translators are registered for the same source/dest - combination, ensure that the lowest-cost one is always inserted - earlier in the list ........ r46153 | kpfleming | 2006-10-24 - 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines code zone experiment: - don't offer formats in the outbound INVITE that aren't either - passthrough or translatable ........ - - * channels/chan_iax2.c: restore bugfix that was reverted by - trunk_mtu patch - - * channels/chan_sip.c, /, apps/app_record.c, apps/app_softhangup.c, - res/res_adsi.c, main/utils.c, pbx/dundi-parser.c, - apps/app_ices.c, apps/app_getcpeid.c, apps/app_queue.c, - channels/chan_iax2.c, main/cli.c, main/cdr.c, - channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c, - channels/chan_h323.c, pbx/pbx_ael.c, channels/chan_alsa.c, - pbx/pbx_realtime.c, apps/app_sms.c, channels/chan_nbs.c, - main/image.c, main/db.c, channels/chan_mgcp.c, cdr/cdr_custom.c, - apps/app_parkandannounce.c, apps/app_voicemail.c: Merged - revisions 46200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) - | 2 lines apparently developers are still not aware that they - should be use ast_copy_string instead of strncpy... fix up many - more users, and fix some bugs in the process ........ - -2006-10-25 14:26 +0000 [r46199] Olle Johansson <oej@edvina.net> - - * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Ok, - second attempt... - -2006-10-25 14:18 +0000 [r46198] Luigi Rizzo <rizzo@icir.org> - - * CHANGES: document a couple of recently introduced feature also - including the version number where the feature appeared. - -2006-10-25 14:14 +0000 [r46183-46197] Olle Johansson <oej@edvina.net> - - * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: On the - other hand, don't use 1.4 patches for trunk... Sorry. - - * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Add - ability to adapt the IAX trunk packets to the MTU size, to avoid - bad audio when the number of channels fill the MTU on a given - link. In the future, this needs to be configurable per peer with - trunking enabled. - - * channels/chan_sip.c: Adding comments in the source is more - persistent than just adding them to the commit message :-) - - * channels/chan_sip.c: Always add doxygen comments to new - functions, more lines than one are appreciated really. (Read the - coding guidelines). I've worked hard to make chan_sip a better - place to code in, let's keep it that way and don't add more stuff - without comments. Thank you. - -2006-10-25 00:32 +0000 [r46155] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c, /, main/translate.c, formats/format_pcm.c, - channels/chan_h323.c, channels/chan_iax2.c, - include/asterisk/frame.h, main/rtp.c: Merged revisions 46154 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) - | 2 lines add passthrough and file format support for G.722 16KHz - audio (issue #5084, original patch by andrew, updated by - mithraen) ........ - -2006-10-24 20:22 +0000 [r46141] Mark Spencer <markster@digium.com> - - * res/res_agi.c: Fix FastAGI to not wait for the non-existant pid - -2006-10-24 19:33 +0000 [r46131] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 46130 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46130 | file | 2006-10-24 15:29:56 -0400 (Tue, 24 Oct 2006) | 2 - lines We need to initialize our scheduler pthread condition... - yes. ........ - -2006-10-24 17:14 +0000 [r46104-46120] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: i really think it is safe to commit this version, - that simplifies the manager queue handling as described in the - comment, and will make a lot easier to make further work on this - code. - - * channels/chan_sip.c: correct fix for the bug i previously - introduced - the strings are meant to be always initialized, - independently from their content. - -2006-10-24 05:24 +0000 [r46094] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 46093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46093 | russell | 2006-10-24 01:23:33 -0400 (Tue, 24 Oct 2006) | - 3 lines Restore the ability to remove the firmware directory - without causing the installation to fail (issue #8111) ........ - -2006-10-24 03:15 +0000 [r46081] Kevin P. Fleming <kpfleming@digium.com> - - * doc/imapstorage.txt, /: Merged revisions 46080 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46080 | kpfleming | 2006-10-23 22:13:08 -0500 (Mon, 23 Oct 2006) - | 2 lines simplify and correct voicemail IMAP storage build - instructions ........ - -2006-10-24 03:09 +0000 [r46079] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 46078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006) - | 3 lines Pass through a frame if we don't know what it is, - rather than trying to pass a NULL, which will segfault a channel - driver (Bug 8149) ........ - -2006-10-24 01:28 +0000 [r46055-46068] Russell Bryant <russell@digium.com> - - * utils/muted.c, /, utils/ael_main.c: Merged revisions 46067 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46067 | russell | 2006-10-23 21:27:42 -0400 (Mon, 23 Oct 2006) | - 7 lines In muted.c, check the return value of strdup. In - ael_main.c, check the return value of calloc. (issue #8157) In - passing fix a few minor bugs in ael_main.c. The last argument to - strncpy() was a hard-coded 100, where it should have been 99. I - changed this to use sizeof() - 1. ........ - - * /, apps/app_meetme.c: Merged revisions 46065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r46065 | russell | 2006-10-23 21:04:14 -0400 (Mon, 23 Oct 2006) | - 2 lines Fix the descriptions of some of the MeetMeAdmin options - (issue #8098, mflorell) ........ - - * channels/chan_sip.c: Fix a seg fault on a registration. Line - 7706, in parse_register_contact, explicitly passes NULL as the - "pass" argument to this function. - -2006-10-23 21:46 +0000 [r46003-46045] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: Unlike ast_strdup(), ast_strdupa() does not - take a NULL pointer as argument, so fix the places where this - might happen. This is also a fix that ought to go into 1.4 [The - difference between the two functions is a bit confusing, and in - asterisk i believe all string handling functions should be able - to handl a NULL string as argument, but changing the API in trunk - and not in 1.4 would make backporting harder.] - - * channels/chan_sip.c: remove a useless check for ocseq = 0. As - discussed on the mailing lists, 0 is a legal value for Cseq, so - there is no point to treat it specially. - - * channels/chan_sip.c: get_header() always returns a non-NULL - value, so checking for NULL is certainly wrong and usually - disables the checks that we want to make instead. This commit - fixes a number of the above bugs where the result of get_header() - is immediately checked for NULL. This is certainly a candidate - for merging into 1.4 - - * channels/chan_sip.c: put another duplicated block of code in a - function. - - * channels/chan_sip.c: reformat a statement and comment a - potentially wrong assignement (altering state on an unvalidated - message). - - * channels/chan_sip.c: Remove unnecessary casts from const char * - to char *, if necessary by slightly rearranging the code. - - * channels/chan_sip.c: another use for parse_uri(). On passing, - remove a wrong comment (that probably I wrote myself!) and - introduce a temporary variable to avoid a misleading cast. - -2006-10-23 17:08 +0000 [r46000] Russell Bryant <russell@digium.com> - - * /, res/res_jabber.c: Merged revisions 45999 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45999 | russell | 2006-10-23 13:07:45 -0400 (Mon, 23 Oct 2006) | - 2 lines don't crash when an incoming message has no "from" (issue - #8205, jmls) ........ - -2006-10-23 16:54 +0000 [r45945-45989] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c: use autodetected support for gethostbyname_r - - * channels/chan_sip.c: + make sure parse_uri never returns NULL - pointers - this simplifies its usage. + add another client for - parse_uri, in handling Contact: strings (on passing, document the - content of the "fullcontact" field); + in register_verify(), mark - with XXX what i believe is another misinterpretation on the URI - format when '@' is missing. No code changed here, so no fixes - applied. - - * channels/chan_sip.c: After reading better the SIP RFC on sip URI - (19.1.1) fix parse_uri() to interpret a missing userinfo section - as a domain-only URI, and comment a wrong interpretation of the - above in check_user_full(). The function has been patched to - preserve the existing behaviour (in what admittedly is a corner - case, but could be received under attacks). Hopefully the From: - based matching will go away soon! - - * channels/chan_sip.c: in function get_also_info(), move argument - stripping before splitting around the @, otherwise the - refer_to_domain might contain arguments as well, causing - failures. I think this is a true bug that ought to be fixed in - 1.4 as well. - - * channels/chan_sip.c: start putting the URI parsing code in one - place, introducing the function parse_uri() that splits a URI in - its components. Right now use it only in one place, because the - custom parsing that is done here and there sometimes has bugs - that i want to figure out first. - - * channels/chan_sip.c: put common code in function terminate_uri() - so we need to fix it only in one place. - - * channels/chan_sip.c: More cleanup of check_user_full with no - functional change apart from a small (but disabled by default) - new option. In detail: + introduce a new value for enum - check_auth_result, AUTH_DONT_KNOW, used (read below) when a - function does not have a conclusive response. Possibly this is - the same as AUTH_NOT_FOUND, but need to check further. + move the - large blocks (checking in the users list and in the peers list, - respectively) from check_user_full() to separate functions. They - return AUTH_DONT_KNOW in case they don't find a match, so the - caller know that it has to try the next method. There is still - some duplication of code here, but i have not tried yet to remove - it. + [new option] a new option in sip.conf, match_auth_username, - has been introduced, and disabled by default. If set, and the - incoming request carries authentication info, the username to - match in the users list is taken from there rather than from the - From: field. This change is easy to identify, being made of - one - line to declare the variable match_auth_username - a block of 15 - lines in check_user_full() - one line in sip list settings - two - lines for parsing the config file. check_user_full() is now a lot - cleaner - basically a sequence of checks that are applied to the - request. This will help future work with new matching schemes. - -2006-10-23 00:33 +0000 [r45929] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_odbc.c: Merged revisions 45928 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45928 | file | 2006-10-22 20:27:39 -0400 (Sun, - 22 Oct 2006) | 10 lines Merged revisions 45927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 - lines Don't leak memory mmmk? ........ ................ - -2006-10-22 21:57 +0000 [r45917] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 45916 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45916 | crichter | 2006-10-22 23:44:46 +0200 - (Sun, 22 Oct 2006) | 9 lines Merged revisions 45808 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 - Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and - couldn't be initialized it would cause a segfault after 'reload'. - Reported by Drew/Matt thx. ........ ................ - -2006-10-22 21:08 +0000 [r45904-45915] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: more streamlining of check_user_full - - * channels/chan_sip.c: simplify the flow of function - check_user_full() A large block needs reindentation now, but we - don't do that because it can be moved to a separate function. - - * channels/chan_sip.c: put duplicated code in functions. - -2006-10-22 19:34 +0000 [r45893] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in: regenerate the - configure script and autoconfig.h.in to reflect recent changes - for https support for the built in http server - -2006-10-22 19:09 +0000 [r45858-45892] Luigi Rizzo <rizzo@icir.org> - - * main/Makefile, configure.ac, main/http.c, - configs/http.conf.sample: Fix a few issues in the previous - (disabled) HTTPS code, and support linux as well (using - fopencookie(), which should be available in glibc). Update - configure.ac to check for funopen (BSD) and fopencookie(glibc), - and while we are at it also for gethostbyname_r (the generated - files need to be updated, or you need to run bootstrap.sh - yourself). Document the new options in http.conf.sample (names - are only tentative, better ones are welcome). At this point we - can safely enable the option. Anyone willing to try this on Sun - and Apple platforms ? - - * main/http.c: Implement https support. The changes are not large. - Most of the diff comes from putting the global variables - describing an accept session into a structure, so we can reuse - the existing code for running multiple accept threads on - different ports. Once this is done, and if your system has the - funopen() library function (and ssl, of course), it is just a - matter of calling the appropriate functions to set up the ssl - connection on the existing socket, and everything works on the - secure channel now. At the moment, the code is disabled because i - have not implemented yet the autoconf code to detect the presence - of funopen(), and add -lssl to main/Makefile if ssl libraries are - present. And a bit of documentation on the http.conf arguments, - too. If you want to manually enable https support, that is very - simple (step 0 1 2 will be eventually detected by ./configure, - the rest is something you will have to do anyways). 0. make sure - your system has funopen(3). FreeBSD does, linux probably does - too, not sure about other systems. 1. uncomment the following - line in main/http.c // #define DO_SSL /* comment in/out if you - want to support ssl */ 2. add -lssl to AST_LIBS in main/Makefile - 3. add the following options to http.conf sslenable=yes - sslbindport=4433 ; pick one you like sslcert=/tmp/foo.pem ; path - to your certificate file. 4. generate a suitable certificate e.g. - (example from mini_httpd's Makefile: openssl req -new -x509 -days - 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem and here you - go: https://localhost:4433/asterisk/manager now works. - - * main/http.c: it is useless and possibly wrong to use ast_cli() to - send the reply back to http clients. Use fprintf/fwrite instead, - since we are already using a FILE * to read the input. If you - wonder why, this is because it makes it trivial to implement - https support (as long as your system has funopen()). And this is - what i am going to put in with the next few commits... - -2006-10-22 04:44 +0000 [r45847] Joshua Colp <jcolp@digium.com> - - * Makefile, main/Makefile: Let's have build.h created a bit earlier - so that func_version can use it and not stop the build on a fresh - machine that has never had Asterisk installed on it before... - -2006-10-21 20:24 +0000 [r45836] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: the default port number was erroneously stored in - host order, and reading from the config file used ntohs instead - of htons. this ought to be merged to 1.4 as well. - -2006-10-21 18:52 +0000 [r45820] Joshua Colp <jcolp@digium.com> - - * /, main/loader.c: Merged revisions 45817 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45817 | file | 2006-10-21 14:48:58 -0400 (Sat, 21 Oct 2006) | 2 - lines Don't use promotion on Darwin because it doesn't seem to - work quite right in all cases, this should solve the unresolved - symbol issue people have been seeing. ........ - -2006-10-21 18:50 +0000 [r45819] Russell Bryant <russell@digium.com> - - * /, res/res_monitor.c: Merged revisions 45818 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45818 | russell | 2006-10-21 14:49:46 -0400 (Sat, 21 Oct 2006) | - 3 lines Add a couple missing unregistrations of manager actions - and remove duplicate unregistrations of applications. (issue - #8194, jmls) ........ - -2006-10-20 20:59 +0000 [r45786] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: introduce sip_pvt_lock() and - sip_pvt_unlock() wrappers to lock these data structures. This - improve readability, and also hides the underlying locking - mechanism so it is a lot easier to add diagnostic code, or move - the object locks somewhere else, etc. On passing, rename the lock - field in sip_pvt to pvt_lock, also for ease of readability. - -2006-10-20 19:04 +0000 [r45776] Joshua Colp <jcolp@digium.com> - - * Makefile, /: Merged revisions 45775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45775 | file | 2006-10-20 15:03:03 -0400 (Fri, 20 Oct 2006) | 2 - lines Pass DESTDIR and ASTSBINDIR so that the utilities get - installed in the proper location (reported on asterisk-dev - mailing list) ........ - -2006-10-20 15:54 +0000 [r45764] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: put the constants for whether methods can - create a dialog or not in an enum - -2006-10-20 11:24 +0000 [r45753] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: minor comment changes, code rearrangement and - field renaming to minimize diffs with future modifications. The - current implementation is problematic for the following reasons: - + all insertions are O(N) because the event list does not have a - tail pointer; + there is only a single lock protecting both - session and users queues. + the implementation of the queue - itself is not documented. I think i have figured it out, more or - less, but am unclear on whether there is proper locking in place - The rewrite (which i have working locally) uses a tailq so - insertions are O(1), separate locks for the event and session - queues, and has a documented implementation so hopefully we can - figure out if/where bug exist. - -2006-10-20 08:14 +0000 [r45742-45743] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Let's repair the SIP attack shield :-) - - * main/manager.c: Doxygen corrections - -2006-10-19 22:06 +0000 [r45712-45724] Steve Murphy <murf@digium.com> - - * funcs/func_version.c (added): This new function, VERSION(), - created via bug report 8176, may help dialplan programmers in the - future. In the meantime, they can use the algorithm I outline on - the bug report notes; If anyone invents something better, I'd - hope they post it - - * utils/astman.c: astman was slightly weirding out over the new - Dial and Newcallerid events - -2006-10-19 17:26 +0000 [r45696] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: more fixes to comments and very minor code - rearrangement. - -2006-10-19 17:25 +0000 [r45693-45695] Joshua Colp <jcolp@digium.com> - - * /, res/res_jabber.c: Merged revisions 45694 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45694 | file | 2006-10-19 13:24:40 -0400 (Thu, 19 Oct 2006) | 2 - lines Let's remember to unregister JabberStatus too (issue #8184 - reported by jmls) ........ - - * /, apps/app_externalivr.c: Merged revisions 45692 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45692 | file | 2006-10-19 13:19:47 -0400 (Thu, - 19 Oct 2006) | 10 lines Merged revisions 45691 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2 - lines Respect language selection when seeing if the file exists - (issue #8178 reported by mnicholson) ........ ................ - -2006-10-19 17:07 +0000 [r45690] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: implement proper XML/HTML formatting of multiple - messages (e.g. the result of waitevent). Also fix some comments. - -2006-10-19 16:06 +0000 [r45679] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 45678 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45678 | file | 2006-10-19 12:03:09 -0400 (Thu, 19 Oct 2006) | 2 - lines If the jitterbuffer is forced on then we can't partially - bridge (reported by wangster on #asterisk-dev) ........ - -2006-10-19 10:05 +0000 [r45648-45668] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: move a large block out of do_monitor() and - into a function, to improve readability. - - * channels/chan_sip.c: + move the definition of netlock as it was - not related to the comment just above; + decouple the struct - definition and variable declaration (iflist); - - * main/manager.c: more documentation of data structure and - functions. Of interest: + ast_get_manager_by_name_locked() is now - without the ast_ prefix as it is a local function; + - unuse_eventqent() renamed to unref_event(), and returns the - pointer to the next entry. + marked with XXX a couple of usages - of unref_event() because i suspect we are addressing the wrong - entry. - -2006-10-19 07:17 +0000 [r45647] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Cleaning up... Removing duplicate (again) - -2006-10-19 02:16 +0000 [r45634] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, include/asterisk/threadstorage.h: restore - freeing of threadstorage objects without custom cleanup functions - allow custom threadstorage init functions to return failure use a - custom init function for chan_sip's temp_pvt, to improve - performance a bit - -2006-10-19 01:04 +0000 [r45623-45624] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merge fix to not leak the stringfields of - a thread speicif sip_pvt. This also includes the fix not to leak - the actual sip_pvt. Merged revisions 45622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45622 | russell | 2006-10-18 20:59:51 -0400 (Wed, 18 Oct 2006) | - 2 lines Don't leak the actual thread-specific sip_pvt struct - ........ - - * main/channel.c, main/frame.c, main/manager.c, - channels/chan_sip.c, channels/chan_skinny.c, main/logger.c, - main/utils.c, channels/iax2-parser.c, - include/asterisk/threadstorage.h, main/cli.c: Extend the thread - storage API such that a custom initialization function can be - called for each thread specific object after they are allocated. - Note that there was already the ability to define a custom - cleanup function. Also, if the custom cleanup function is used, - it *MUST* call free on the thread specific object at the end. - There is no way to have this magically done that I can think of - because the cleanup function registered with the pthread - implementation will only call the function back with a pointer to - the thread specific object, not the parent ast_threadstorage - object. - -2006-10-18 22:40 +0000 [r45611] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: silent warning from a debugging message (which - will go away soon, anyways) - -2006-10-18 22:19 +0000 [r45610] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c, CHANGES: Just for Nicholson - here's an - option, C, to Meetme that will allow it to continue in the - dialplan if the person is kicked out. (issue #7994 reported by - mnicholson with mods by myself) - -2006-10-18 21:41 +0000 [r45597-45599] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: remove trailing whitespace - - * main/manager.c: ouch! remember to unlink temporary files once - done with them. - - * main/manager.c: + move output_format variables in the http - section of the file; + more comments on struct mansession and - global variables; + small improvements to the session matching - code so it supports multiple sessions from the same IP - -2006-10-18 21:05 +0000 [r45596] Joshua Colp <jcolp@digium.com> - - * /, main/asterisk.c: Merged revisions 45595 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45595 | file | 2006-10-18 17:03:34 -0400 (Wed, 18 Oct 2006) | 2 - lines Don't modify things if we are using vfork as this is very - bad and may cause unexpected behavior (issue #7970 reported by - Nick Gavrikov) ........ - -2006-10-18 17:53 +0000 [r45572-45583] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: another bunch of comments on the data structures. - - * main/manager.c: despite the large changes, this commit only moves - functions around so that functions belonging to the same group - are close to each other. At the beginning of each group i have - added a bit of documentation to explain what the group does and - what is the typical flow - basically, all i have learned by code - inspection over the past few days should be documented for you to - read. I have not put many doxygen annotations just because i am - not sure what are the proper ones. Hopefully some doxygen experts - will jump in. Next on the plate: try to figure out how "struct - eventqent" are supposed to work. - - * main/manager.c: more comment and formatting fixes, small - simplifications to functions get_input() and session_do() - -2006-10-18 16:45 +0000 [r45571] Matt O'Gorman <mogorman@digium.com> - - * main/manager.c: rizzo compile then commit, maybe even run it too - ^_^ - -2006-10-18 15:49 +0000 [r45529-45561] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: comment and cleanup the main thread. On passing, - fix a bug: close the socket if the allocation of a structure for - the new session fails. (the bugfix is a candidate for 1.4) - - * main/manager.c: create a new (internal, for the time being) - function astman_start_ack() to start manager responses that need - further lines. This removes a lot of duplicate code from the - various handlers that at the moment build an ActionID string - themselves. Once settled, the function should move to manager.h - so it can be used by other files (chan_agent, chan_iax2, - chan_sip, chan_zap, res_jabber and app_queue). I am not totally - clear if there is a preferred position for the ActionID: line in - a message. Some instances put it at the end, but one would argue - that it is preferable to have it at the beginning. - - * main/manager.c: more indentation cleanup from previous commits, - and remove the "busy" field from struct mansession as it was not - used correctly anyways. - - * main/manager.c: create proper handlers for "Challenge" and - "Login" actions, rather than use inline code for them. Things are - more readable this way, and also error processing is more - consistent. - - * main/manager.c: fix indentation from a commit of a couple of days - ago - - * main/manager.c: another batch of simplifications to - authenticate() (they are committed a bit at a time so it is - easier to revert them in case we find a bug at a later time). - -2006-10-18 12:15 +0000 [r45528] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Remove duplicate declarations... - -2006-10-18 11:59 +0000 [r45463-45518] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c, configs/manager.conf.sample: remove unused fields - and unimplemented options. - - * main/manager.c: first pass as simplifying authenticate(), - avoiding whitespace changes - - * main/manager.c: more code simplifications - - * main/manager.c: simplify ast_strings_to_mask - - * main/manager.c: add a comment to remember that a block of code is - completely redundant. - - * main/manager.c: + move the enum declaration for output formats - near the head of the file, so it can be used from more places; + - make the declaration of contenttype[] more robust; + remove the - wrappers around __xml_translate(), since they were used only in - one place, and rename to xml_translate(). This allows for a bit - of simplifications. + document the output produced by the above - function. - - * main/manager.c: merge xml_translate() and html_translate() into - one function since they do similar things. Add a small form on - top of the html output so request like - http://foo:8088/asterisk/manager will suggest you what to do. - Note: i suspect there is still a bug somewhere in the session - matching code, as sometimes you have to login twice in order for - the following commands to be recognised. Apart from this, the cli - now is basically usable from a web form! - - * main/http.c: introduce uri_decode() so that '+' are translated - into ' ' (e.g. browsers do this when they encode input strings - from a form). - - * main/http.c: various code simplifications to reduce nesting - depth, minor optimizations to avoid extra calls of strlen(), and - some variable localization. One feature worth backporting is the - move of ast_variables_destroy() to a different place in - handle_uri() to avoid leaking memory in case a uri is not found. - -2006-10-18 03:03 +0000 [r45453] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 45452 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2 - lines Don't segfault if you're using a channel driver that - doesn't turn RTCP on ........ - -2006-10-18 02:46 +0000 [r45440-45442] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 45441 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) | - 7 lines Don't attempt to access private data members of the - pthread_mutex_t object, because this does not work on all linux - systems. Instead, just access the reentrancy field in the - ast_mutex_info struct when DEBUG_THREADS is enabled. If - DEBUG_CHANNEL_LOCKS is enabled, the developer probably has - DEBUG_THREADS on as well. (issue #8139, me) ........ - - * configs/sip_notify.conf.sample, /: Merged revisions 45439 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) | - 2 lines update entry to reboot a snom phone (issue #7850, - pnlarsson) ........ - -2006-10-17 23:06 +0000 [r45426] Steve Murphy <murf@digium.com> - - * res/res_agi.c: As per bug 6779, this patch is now applied to - trunk; while I was at it, I corrected a reference to a CLI - command, to follow the new regime. - -2006-10-17 22:32 +0000 [r45409-45411] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/prep_tarball (added): Merged revisions 45410 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45410 | kpfleming | 2006-10-17 17:31:54 -0500 (Tue, 17 Oct 2006) - | 2 lines add a project-specific script to be used during release - preparation ........ - - * main/channel.c, /, channels/chan_sip.c, channels/chan_iax2.c, - include/asterisk/stringfields.h, main/ast_expr2.c: Merged - revisions 45408 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) - | 3 lines optimize the 'quick response' code a bit more... no - more malloc() or memset() for each response expand stringfields - API a bit to allow reusing the stringfield pool on a structure - when needed, and remove some unnecessary code when the structure - was being freed ........ - -2006-10-17 21:09 +0000 [r45379-45398] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Warning be gone! - - * /, channels/chan_sip.c: Merged revisions 45378 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2 - lines Don't create a "real" pvt structure for requests that - shouldn't be able to create one. Instead use a temporary pvt and - fill it with enough information so we can send a reply. ........ - -2006-10-17 19:57 +0000 [r45365] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, doc/channelvariables.txt: Issue #5484 - (branch sipdiversion) - Support for Diversion header in redirects - of calls with 302 redirection. (tinning) - -2006-10-17 18:08 +0000 [r45351] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: simplify authority_to_str() using - ast_build_string() - -2006-10-17 17:54 +0000 [r45335] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7254 - Add support of "423 Interval - too brief" to outbound SIP registrations. Thanks, tardieu! - -2006-10-17 17:51 +0000 [r45334] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: Improve the XML formatting of responses coming - from web interface. Normal responses are sequences of lines of - the form "Name: value", with \r\n as line terminators and an - empty line as a response terminator. Generi CLI commands, - however, do not have such a clean formatting, and the existing - code failed to generate valid XML for them. Obviously we can only - use heuristics here, and we do the following: - accept either \r - or \n as a line terminator, trimming trailing whitespace; - if a - line does not have a ":" in it, assume that from this point on we - have unformatted data, and use "Opaque-data:" as a name; - if a - line does have a ":" in it, the Name field is not always a legal - identifier, so replace non-alphanum characters with underscores; - All the above is to be refined as we improve the formatting of - responses from the CLI. And, all the above ought to go as a - comment in the code rather than just in a commit message... - -2006-10-17 17:51 +0000 [r45331-45333] Olle Johansson <oej@edvina.net> - - * /, configs/sip.conf.sample: Update of docs - - * channels/chan_sip.c: - Remove unneeded code that won't be reached - now that we kill responses to unkonwn dialogs earlier in the - process. - move debug message. - -2006-10-17 17:41 +0000 [r45330] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: open a temporary file to receive the output from - cli commands invoked through the http interface. It is not - terribly efficient but better than no output at all. Todo: use a - configurable /tmp directory instead of a hardwired one. - -2006-10-17 17:22 +0000 [r45328] Kevin P. Fleming <kpfleming@digium.com> - - * /, LICENSE: Merged revisions 45327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45327 | kpfleming | 2006-10-17 12:22:25 -0500 - (Tue, 17 Oct 2006) | 10 lines Merged revisions 45326 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45326 | kpfleming | 2006-10-17 12:22:01 -0500 (Tue, 17 - Oct 2006) | 2 lines provide licensing language for IAXy firmware - file ........ ................ - -2006-10-17 17:19 +0000 [r45325] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: document xml_copy_escape() and add an extra - function, namely replace non-alphanum chars with underscore. This - is useful when building field names in xml formatting. - -2006-10-17 16:27 +0000 [r45295-45316] Olle Johansson <oej@edvina.net> - - * /: ...block this one too... Only applies to 1.4 since the fix for - trunk was different. - - * /: Block patch from 1.4 that does not apply here. - - * channels/chan_sip.c: Get rid of the ignore variable that was only - partially replaced by the flag. - -2006-10-16 20:26 +0000 [r45234-45286] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: In the course of a - data this has been turned into an option to ignore replies, then - ignore responses and finally I'm just getting rid of the option - altogether and making it the default no matter what. C'est la - vie! - - * /: Woof. - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 45280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, - 16 Oct 2006) | 10 lines Merged revisions 45265 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 - lines Use responses rather then replies even though they mean the - same thing. ........ ................ - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 45262 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, - 16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 - lines Add 'ignoreoodreplies' option which will not create a pvt - structure on a SIP response but instead basically drop it. - ........ ................ - - * apps/app_directed_pickup.c: It's new directed pickup! This now - features a more sane way of finding the channel to pick up (I - snuck it into the tree on Friday... bet you didn't know I'd - actually use it eh?). PICKUPMARK now also works in a different - way, you should prefix it with _ when setting it so it gets - inherited onto the channel(s) created in app_dial as directed - pickup will now look for it on the target channel, not the - originating channel. (BE-85) - -2006-10-16 14:03 +0000 [r45224] Olle Johansson <oej@edvina.net> - - * CREDITS, /: Update - -2006-10-16 14:00 +0000 [r45219] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: + comment some unclear fields of struct - mansession; + let some commands (Challenge, Login) be processed - even if already authenticated, as it doesn't harm and prevents - some incorrect error messages + remove custom code for Logoff - - the existing handler was ok. Some indentation fixes may be - necessary - -2006-10-16 13:20 +0000 [r45194-45209] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: When adding new functions, please add a - forward declaration. I *know* it is not required, but it makes - navigation easier and will help when splitting up this large - source code file. Thank you! - - * /, channels/chan_sip.c: Importing rev 45196 from 1.4 - don't kill - dialog for a bad response - - * channels/chan_sip.c: A B2BUA should *not* issue proxy auth. - -2006-10-16 11:29 +0000 [r45151-45185] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: + comment some unclear requirements for - master_eventq + remove the need for an snprintf in - astman_get_header() + fix comment for manager list eventq + - localize one variable and minor code simplifications. - - * main/manager.c: protect access to first_action with actionlock. - Mark with XXX one place (during command execution) where - navigation should be protected with actionlock, but is not - because it would block requests for a long time. To solve this - properly we need to put reference counts in the struct - manager_action. A suboptimal fix is to copy the record on a - search and then unlock the list while we work on the copy. - - * main/http.c: comment some functions, and more small code - simplifications - - * main/http.c: fix indentation of a large block after changes in - previous commit (basically whitespace only). - - * main/http.c: simplify string parsing routines using ast_skip_*() - functions. - - * main/http.c: don't forget to close a descriptor on a malloc - failure. On passing, small rearrangement of the code to reduce - indentation. There is a bit more cleanup planned for this file, - so a merge to 1.4 will be done when it is all done. - - * main/http.c: typo: serer -> server - -2006-10-14 04:36 +0000 [r45142] Steve Murphy <murf@digium.com> - - * funcs/func_rand.c: update the doc string for both AEL and - extensions.conf users. - -2006-10-13 23:03 +0000 [r45126] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/acl.c: Merged revisions 45125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006) - | 7 lines - ------------------------------------------------------------------------ - r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006) - | 2 lines don't drop the entire permit/deny list when an attempt - is made to add an invalid entry (BE-92) - ------------------------------------------------------------------------ - ........ - -2006-10-13 21:20 +0000 [r45105-45109] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Inherit the context and extension until the - channel is answered - - * /, res/res_speech.c: Merged revisions 45106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45106 | file | 2006-10-13 17:06:09 -0400 (Fri, 13 Oct 2006) | 2 - lines Clear the quiet flag too since we are restarting a - recognition again (reported on -dev by Stephan Edelman) ........ - - * /, res/res_speech.c: Merged revisions 45104 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45104 | file | 2006-10-13 17:01:13 -0400 (Fri, 13 Oct 2006) | 2 - lines Check return value from engine in case of failure (ie: out - of licenses) (reported on -dev mailing list) ........ - -2006-10-13 19:24 +0000 [r45089] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 45088 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r45088 | crichter | 2006-10-13 21:19:46 +0200 (Fr, 13 - Okt 2006) | 1 line avoiding warning, fixing potential bug - ........ - -2006-10-13 18:45 +0000 [r45080] Joshua Colp <jcolp@digium.com> - - * codecs/lpc10/median.c, codecs/lpc10/encode.c, - codecs/lpc10/ivfilt.c, /, codecs/lpc10/bsynz.c, - codecs/lpc10/prepro.c, codecs/lpc10/invert.c, - codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, - codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, - codecs/lpc10/pitsyn.c, codecs/lpc10/difmag.c, - codecs/lpc10/voicin.c, codecs/lpc10/synths.c, - codecs/lpc10/preemp.c, codecs/lpc10/hp100.c, - codecs/lpc10/lpfilt.c, codecs/lpc10/rcchk.c, - codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, - codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, - codecs/lpc10/lpcini.c, codecs/lpc10/random.c, - codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, - codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, - codecs/lpc10/analys.c, codecs/lpc10/onset.c, - codecs/lpc10/energy.c, codecs/lpc10/lpcdec.c, - codecs/lpc10/deemp.c: Merged revisions 45079 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45079 | file | 2006-10-13 14:42:49 -0400 (Fri, 13 Oct 2006) | 2 - lines And file said... let the compiler warnings STOP! ........ - -2006-10-13 18:08 +0000 [r45078] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest17 (added), - pbx/ael/ael-test/ael-vtest17/extensions.ael (added), - pbx/ael/ael-test/ael-vtest17 (added), - pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Correction for bug - 8128 in trunk - -2006-10-13 17:06 +0000 [r45052-45067] Joshua Colp <jcolp@digium.com> - - * /, apps/app_chanspy.c: Merged revisions 45066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45066 | file | 2006-10-13 13:05:02 -0400 (Fri, - 13 Oct 2006) | 10 lines Merged revisions 45060 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45060 | file | 2006-10-13 13:01:22 -0400 (Fri, 13 Oct 2006) | 2 - lines Turn on volume adjustment if it needs to be on (issue #8136 - reported by mnicholson) ........ ................ - - * /, apps/app_playback.c: Merged revisions 45051 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45051 | file | 2006-10-13 12:20:58 -0400 (Fri, 13 Oct 2006) | 2 - lines Move say.conf existence check to do_say function since it - is called from multiple places (issue #8144 reported by kshumard) - ........ - -2006-10-13 16:20 +0000 [r45050] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 45049 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45049 | kpfleming | 2006-10-13 11:19:35 -0500 - (Fri, 13 Oct 2006) | 10 lines Merged revisions 45048 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45048 | kpfleming | 2006-10-13 11:18:08 -0500 (Fri, 13 - Oct 2006) | 2 lines when sending a call to a peer, use the proper - socket if we have multiple bindings (reported on asterisk-dev) - ........ ................ - -2006-10-13 16:02 +0000 [r45032-45047] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 45040 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45040 | file | 2006-10-13 12:01:17 -0400 (Fri, 13 Oct 2006) | 2 - lines Complete merging in RPID screen changes (issue #8101 - reported by hristo, patch by oej in revision 44757) ........ - - * main/dnsmgr.c, /: Merged revisions 45031 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45031 | file | 2006-10-13 11:53:22 -0400 (Fri, - 13 Oct 2006) | 10 lines Merged revisions 45030 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45030 | file | 2006-10-13 11:49:53 -0400 (Fri, 13 Oct 2006) | 2 - lines Pass the right value to usleep for sleeping, and always add - the background refresh item back into the scheduler if enabled - since it is deleted during reload. (issue #8142 reported by - p_lindheimer) ........ ................ - -2006-10-13 15:47 +0000 [r45029] Kevin P. Fleming <kpfleming@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: Merged revisions 45027 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r45027 | kpfleming | 2006-10-13 10:41:14 -0500 (Fri, 13 Oct 2006) - | 2 lines use a configure script test for PMTU discovery control - instead of just assuming it's available on Linux ........ - -2006-10-13 15:42 +0000 [r45028] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 45026 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r45026 | crichter | 2006-10-13 16:45:39 +0200 - (Fr, 13 Okt 2006) | 9 lines Merged revisions 45020 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45020 | crichter | 2006-10-13 15:11:13 +0200 (Fr, 13 - Okt 2006) | 1 line fixed some echocandisable issues when bridged. - this caused a kernel panic sometimes..also some minor formatting - fixes ........ ................ - -2006-10-13 11:18 +0000 [r45009-45010] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: Try to avoid the use of 'z' modifier in - cases where it is not necessary - rather, cast the argument to - int. In this case, the string is in a UDP packet and as such - limited to 64k so its length can be safely represented in an int - without truncation (besides, this is just a debugging message!) - - * channels/chan_sip.c: arguments to auth_headers() needed to be - swapped here. To avoid the same mistake in the future (due to - slightly confusing variable names), add a comment. On passing, - remove a redundant initialization. - -2006-10-13 08:23 +0000 [r45000] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/isdn_msg_parser.c: Merged revisions 44994 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44994 | crichter | 2006-10-13 09:52:41 +0200 - (Fr, 13 Okt 2006) | 9 lines Merged revisions 44993 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44993 | crichter | 2006-10-13 09:40:07 +0200 (Fr, 13 - Okt 2006) | 1 line fixed issue, that the hangupcause got a wrong - isdn cause at RELEASE_COMPLETE ........ ................ - -2006-10-12 20:41 +0000 [r44983] Matt O'Gorman <mogorman@digium.com> - - * /, channels/chan_gtalk.c: Merged revisions 44982 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.4 ........ - r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006) - | 2 lines fix for bug 7764. ........ - -2006-10-12 19:16 +0000 [r44957-44973] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: eliminate compiler warning - - * /, channels/chan_sip.c: Merged revisions 44971 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44971 | kpfleming | 2006-10-12 14:14:24 -0500 (Thu, 12 Oct 2006) - | 2 lines we can only send one 'a=ptime' attribute per media - session, not one for each format ........ - - * include/asterisk/utils.h, /, channels/chan_sip.c, main/utils.c, - main/netsock.c: Merged revisions 44956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44956 | kpfleming | 2006-10-12 13:38:51 -0500 - (Thu, 12 Oct 2006) | 10 lines Merged revisions 44955 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12 - Oct 2006) | 2 lines ensure that IAX2 and SIP sockets allow UDP - fragmentation when running on Linux (thanks to Brian Candler on - the asterisk-dev list for the tip) ........ ................ - -2006-10-12 16:57 +0000 [r44944-44946] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 44945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44945 | russell | 2006-10-12 12:56:32 -0400 (Thu, 12 Oct 2006) | - 2 lines fix a silly typo in a comment that I saw while reading - the commit list ........ - - * pbx/pbx_dundi.c: put flags in an enum and remove a couple of - unused defines - -2006-10-12 16:11 +0000 [r44943] Joshua Colp <jcolp@digium.com> - - * Makefile, /: Merged revisions 44942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44942 | file | 2006-10-12 12:08:50 -0400 (Thu, 12 Oct 2006) | 2 - lines Pass off AUDIO_LIBS so muted can link on OSX (issue #8135 - reported by ssokol) ........ - -2006-10-12 15:12 +0000 [r44933] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: + move [almost] all instances of - WWW-Authenticate/Proxy-Authenticate and friends in a function, - auth_headers(), which is used to simplify the interface of - do_{proxy|register}_auth(). + use PROXY_AUTH = 407, WWW_AUTH = - 401 as values for enum sip_auth_type; No functional change, only - code cleanup. - -2006-10-12 13:04 +0000 [r44922] Nadi Sarrar <ns@beronet.com> - - * main/manager.c, /: Merged revisions 44921 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44921 | nadi | 2006-10-12 14:55:25 +0200 (Do, 12 Okt 2006) | 2 - lines append_event must be called while holding the session lock - ........ - -2006-10-12 10:26 +0000 [r44912] Russell Bryant <russell@digium.com> - - * /, res/res_jabber.c: Merged revisions 44911 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44911 | russell | 2006-10-12 06:24:36 -0400 (Thu, 12 Oct 2006) | - 2 lines change some debug output to use LOG_DEBUG instead of - verbose output ........ - -2006-10-11 23:36 +0000 [r44900-44901] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: reduce indentation of two large blocks - - * channels/chan_sip.c: operator != also works between booleans... - -2006-10-11 16:57 +0000 [r44889] Jason Parker <jparker@digium.com> - - * /, main/db1-ast/Makefile: Merged revisions 44888 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r44888 | qwell | 2006-10-11 11:57:06 -0500 (Wed, 11 Oct - 2006) | 3 lines These are already set by the parent Makefile.. - There is no need to have this here (it doesn't actually work - anyways). ........ - -2006-10-11 13:45 +0000 [r44876-44877] Russell Bryant <russell@digium.com> - - * doc/linkedlists.txt (removed): Remove doc/linkedlists.txt as it - is no longer needed. The top of the file reads: As of 2004-12-23, - this documentation is no longer maintained. The doxygen - documentation generated from linkedlists.h should be referred to - in its place, as it is more complete and better maintained. - - * channels/chan_sip.c: Revert Luigi's accidental commit of his - local changes when debugging some SIP authentication issues. This - was committed in revision 44844, where the commit message was - just "small formatting cleanup", so I am pretty sure he didn't - mean to commit this part. - -2006-10-11 13:21 +0000 [r44844-44875] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: remove duplicate prototypes. Have not - checked if there are more. - - * channels/chan_sip.c: simplify and comment - handle_response_peerpoke() - - * channels/chan_sip.c: fix indentation of a function after previous - commit (whitespace-only change) - - * channels/chan_sip.c: handle_response_peerpoke() does not need to - return anything. (Reindentation in the next commit.) - - * channels/chan_sip.c: small formatting cleanup - -2006-10-11 08:45 +0000 [r44840-44843] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 44563 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44563 | crichter | 2006-10-06 14:53:41 +0200 - (Fr, 06 Okt 2006) | 9 lines Merged revisions 44460 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44460 | crichter | 2006-10-05 12:02:38 +0200 (Do, 05 - Okt 2006) | 1 line fixed segfault which happens during - hold/transfer action ........ ................ - - * channels/chan_misdn.c, /: Merged revisions 44562 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44562 | crichter | 2006-10-06 14:52:01 +0200 - (Fr, 06 Okt 2006) | 9 lines Merged revisions 44335 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44335 | crichter | 2006-10-04 17:26:59 +0200 (Mi, 04 - Okt 2006) | 1 line if INFORMATION Message come with keypad - instead of called party number, we just use the keypad as called - party number. ........ ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, - channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged - revisions 44561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44561 | crichter | 2006-10-06 14:50:25 +0200 - (Fr, 06 Okt 2006) | 9 lines Merged revisions 44334 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 - Okt 2006) | 1 line added the option 'reject_cause' to make it - possible to set the RELEASE_COMPLETE - cause on the 3. incoming - PMP channel, which is automatically rejected because chan_misdn - does not support that kind of callwaiting. Therefore chan_misdn - supports now 3 incoming channels on a PMP BRI Port. - misdn_lib_get_free_bc now gets the info if the requested channel - is incoming or outgoing to make the 3. channel possible ........ - ................ - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_lib.c: Merged revisions 44559 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44559 | crichter | 2006-10-06 12:44:34 +0200 - (Fr, 06 Okt 2006) | 9 lines Merged revisions 44149 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44149 | crichter | 2006-10-02 15:28:14 +0200 (Mo, 02 - Okt 2006) | 1 line fixed the hold/retrieve/transfer issues, - removed a useless bc field, added setting of frame.delivery - fields, some minor code cleanups ........ ................ - -2006-10-10 20:52 +0000 [r44831] Tilghman Lesher <tlesher@digium.com> - - * apps/app_rpt.c: More whitespace fixes - -2006-10-10 17:23 +0000 [r44820] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44819 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44819 | file | 2006-10-10 13:21:44 -0400 (Tue, 10 Oct 2006) | 2 - lines Move some stuff around so that a NOTIFY dialog won't hang - around until the end of the world under certain circumstances - ........ - -2006-10-10 16:46 +0000 [r44810] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_logic.c: Merged revisions 44808 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44808 | tilghman | 2006-10-10 11:42:19 -0500 (Tue, 10 Oct 2006) - | 2 lines Lost of a bit of logic when this was simplified between - 1.2 and 1.4 (Bug 8117) ........ - -2006-10-10 16:31 +0000 [r44789-44807] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44806 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44806 | file | 2006-10-10 12:30:00 -0400 (Tue, 10 Oct 2006) | 2 - lines Bail out if we have no refer structure and we get a refer - response ........ - - * /, channels/chan_sip.c: Merged revisions 44788 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44788 | file | 2006-10-10 11:23:14 -0400 (Tue, 10 Oct 2006) | 2 - lines Only set DTMF information if an RTP structure exists - ........ - -2006-10-10 14:54 +0000 [r44787] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged - revisions 44786 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44786 | crichter | 2006-10-10 15:50:26 +0200 - (Di, 10 Okt 2006) | 9 lines Merged revisions 44785 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44785 | crichter | 2006-10-10 15:34:33 +0200 (Di, 10 - Okt 2006) | 1 line (re)added support of dynamical enabling hdlc - on bchannels ........ ................ - -2006-10-10 08:08 +0000 [r44770-44774] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: clarify the use of the standard SIP port - number, 5060, and rename the old DEFAULT_SIP_PORT as - STANDARD_SIP_PORT to make it clear that this is not something we - can change, unlike other defaults. - - * channels/chan_sip.c: improve formatting of SIP packets when - dumped to the verbose output stream, so it is easier to find them - in the log. - -2006-10-09 18:23 +0000 [r44768] Joshua Colp <jcolp@digium.com> - - * funcs/func_timeout.c: Timeout values are in seconds (issue #7122 - reported by jmls) - -2006-10-09 16:15 +0000 [r44765] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 44764 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct - 2006) | 4 lines Fix a problem where phones that go "missing" - never got unregistered. Issue #8067, reported by pj, patch by - Anthony LaMantia (with minor whitespace modifications) ........ - -2006-10-09 15:52 +0000 [r44762-44763] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 44759 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44759 | file | 2006-10-09 11:41:28 -0400 (Mon, 09 Oct 2006) | 2 - lines Properly avoid a collision with iax2_hangup (issue #8115 - reported by vazir) ........ - -2006-10-09 11:20 +0000 [r44753] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Being pedantic... "media" is easier to - understand than "data" in the function name... :-) - -2006-10-09 09:04 +0000 [r44745-44752] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: slightly restructure sipsock_read() removing - a "goto" - - * channels/chan_sip.c: use S_OR in one place - - * channels/chan_sip.c: update_call_counter(): indentation fixes and - small simplifications at the top of the function. - - * channels/chan_sip.c: localize some variables and reduce nesting - depth (indentation will be fixed by a separate commit). - - * channels/chan_sip.c: small simplification to initreqprep() - - * channels/chan_sip.c: Simplify function parse_request() using a - single loop instead of two very similar ones. - - * channels/chan_sip.c: do not dereference p if we know it is NULL. - -2006-10-07 20:42 +0000 [r44697-44731] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fix some debug output for setsockopt for TOS - - * channels/chan_sip.c: - move definition of global_autoframing to - the same place as other globals - set initial value at - load/reload - Add questionmarks for someone to fill in for - doxygen - - * channels/chan_sip.c: Add/change doxygen and comments - - * configs/sip.conf.sample: Recommend using "sip reload" since it's - much easier to learn and remember. - - * channels/chan_sip.c: Explain usage of DEFAULT_SIP_PORT - - * channels/chan_sip.c: Do *NOT* use DEFAULT_SIP_PORT in these - comparisions, since users may change that, but the protocol - clearly states that if we DO NOT mention a port it is 5060. - DEFAULT_SIP_PORT is whatever we default to listen to. I believe - it's the third time I revert a patch like this. - -2006-10-07 14:48 +0000 [r44685-44686] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged - revisions 44684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44684 | pcadach | 2006-10-07 20:39:34 +0600 (Сбт, 07 Окт 2006) | - 1 line Propagate caller's transfer capability too ........ - - * include/asterisk/callerid.h, main/callerid.c, CHANGES, - funcs/func_callerid.c: Extend CALLERID() function for "pres" and - "ton" values - -2006-10-07 12:50 +0000 [r44641-44675] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: slightly restructure the code that computes - the channel's name - - * channels/chan_sip.c: put repeated code to set nat mode in a - function. - - * channels/chan_sip.c: put common code in a function to avoid - repetitions. - - * channels/chan_sip.c: remove hardwired usage of 5060, use - DEFAULT_SIP_PORT instead - - * channels/chan_sip.c: improve and document function - get_in_brackets(), introducing a helper function - find_closing_quote() of more general use. - - * channels/chan_sip.c: when possible, use ast_set2_flags instead of - ast_set/ast_clr . Also mark XXX some dubious places. - -2006-10-06 21:29 +0000 [r44632] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 44631 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44631 | kpfleming | 2006-10-06 16:28:03 -0500 (Fri, 06 Oct 2006) - | 2 lines ensure that mutex locks inside list heads are - initialized properly on platforms that require constructor - initialization (issue #8029, patch from timrobbins) ........ - -2006-10-06 21:10 +0000 [r44630] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 44628 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2 - lines Remove the seqno check for RFC2833, the handler is smart - enough to not need it. ........ - -2006-10-06 21:04 +0000 [r44616-44626] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c: basically fix indentation of a large function - after previous simplifications. On passing, use a single exit - point. (once done with the cleanup i will merge the changes into - 1.4, if applicable) - - * main/manager.c: s cannot be null here, so remove the useless test - and error-handling block. - - * main/manager.c: simplify logic in preparation to reduce - indentation - -2006-10-06 18:47 +0000 [r44606] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: Merged revisions 44605 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2 - lines When the sequence number rolls over then reset the recorded - sequence number for DTMF (issue #8106 reported by bungalow) - ........ - -2006-10-06 17:27 +0000 [r44595] Tilghman Lesher <tlesher@digium.com> - - * apps/app_rpt.c: Massive cleanup of the rpt code, updating to - current coding guidelines - -2006-10-06 16:56 +0000 [r44582] Joshua Colp <jcolp@digium.com> - - * /, main/file.c: Merged revisions 44581 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44581 | file | 2006-10-06 12:53:48 -0400 (Fri, - 06 Oct 2006) | 10 lines Merged revisions 44580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r44580 | file | 2006-10-06 12:52:14 -0400 (Fri, 06 Oct 2006) | 2 - lines Even more frames to treat as though the remote side - disappeared (issue #8097 reported by eldadran) ........ - ................ - -2006-10-06 16:43 +0000 [r44566-44579] Luigi Rizzo <rizzo@icir.org> - - * configs/sip.conf.sample: document a bit the use of templates. - They are highly convenient for writing configuration files, - especially if you have many similar entries, or want to switch - quickly between different configurations without having to - comment/uncomment large sections of the files. - - * configs/sip.conf.sample: document the "contact" option a bit - better. - - * res/res_limit.c: help old bsd-system which don't have RLIMIT_AS - and use RLIMIT_VMEM for virtual memory limits. - - * main/manager.c, main/http.c: make sure sockets are blocking when - they should be blocking. - - * channels/chan_sip.c, configs/sip.conf.sample: Two things: 1. - slightly rearrange/simplify the parsing of the argument in - sip_register. This brings in a patch that has been in Mantis - (5834) for ages, and is the larger part of the commit; 2. - implement the "contact" option for peers, similar to the one in - users.conf: If you put a "contact" option with a non-empty - argument (e.g. contact=123) in a peer section, asterisk will - register with the provider as if you had a register= - username:secret@host/contact line in the general section. The - latter is a very small is a new feature so i am not putting it in - the 1.4 branch, although the "contact" option in user.conf is - already in the 1.4 branch and so it wouldn't be too strange to - merge it. Note that the implementation of "contact" is much - simpler than the one in 5834, and limited to a few lines in - build_peer(). - -2006-10-06 09:01 +0000 [r44545] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove deprecated "incominglimit" config - option - -2006-10-06 06:43 +0000 [r44537] Luigi Rizzo <rizzo@icir.org> - - * configs/sip.conf.sample: update example commands to match current - syntax (does not apply to 1.4) - -2006-10-06 02:24 +0000 [r44527] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in: regenerate the - configure script to reflect the latest changes done by Luigi - Rizzo - -2006-10-05 20:13 +0000 [r44503-44516] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Fix indenting a bit (issue #8082 reported - by selsky) - - * /, main/file.c: Merged revisions 44502 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44502 | file | 2006-10-05 15:57:16 -0400 (Thu, - 05 Oct 2006) | 10 lines Merged revisions 44501 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r44501 | file | 2006-10-05 15:55:41 -0400 (Thu, 05 Oct 2006) | 2 - lines Treat busy control frames as hangup in the file streaming - core (issue #8097 reported by eldadran) ........ ................ - -2006-10-05 18:29 +0000 [r44489] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: These mods fix a problem pointed out by dgartang, - where in certain situations, the target of a goto cannot be - found, even right under your nose. This is because the current - context is not updated properly, and rather than waste time and - find why and where the context should have been updated, I just - use my newly added 'dad' ptrs, and pop until I have either the - context or extension, and use that instead. - -2006-10-05 18:03 +0000 [r44487] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44486 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44486 | file | 2006-10-05 14:01:51 -0400 (Thu, 05 Oct 2006) | 2 - lines One more T.38 fix! Don't leave a reinvite hanging by a - thread if the other side is already setup with T.38 ........ - -2006-10-05 16:11 +0000 [r44477] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/app.c: Merged revisions 44476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44476 | kpfleming | 2006-10-05 11:10:01 -0500 (Thu, 05 Oct 2006) - | 3 lines don't segfault when an argument without a close - parenthesis is found stop parsing as soon as that situation - occurs ........ - -2006-10-05 15:42 +0000 [r44467] Luigi Rizzo <rizzo@icir.org> - - * configure.ac, acinclude.m4: Basically, this commit only - simplifies configure.ac and makes the mechanism more flexible, - but otherwise should not affect your build even if you regenerate - the "configure" script. (Most likely you need to run bootstrap.sh - as you really need to re-run autoheader for reasons that i do not - completely understand). If you don't regenerate "configure", of - course you will see no difference. In detail: - restructure the - check for mandatory modules to remove some redundant code blocks; - - extend the AST_EXT_LIB_CHECK so that it can used also for - checking headers; - define the AST_C_DEFINE_CHECK macro to test - for #defined symbols; - for the two above macros, add a last - argument that getscopied into HAVE_$1_VERSION so the source can - adapt to different versions of the same libraries/header/etc - - document the above; - document a problem that existed before and - i did not manage to solve: the 'description' argument to - AC_DEFINE does not substiture shell variables so you will not see - the actual values in the comments (in autoconfig.h).. - -2006-10-05 02:43 +0000 [r44451] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2 - lines Don't totally bail out if T.38 was negotiated ........ - -2006-10-05 01:43 +0000 [r44437] Kevin P. Fleming <kpfleming@digium.com> - - * utils/Makefile, /: Merged revisions 44436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44436 | kpfleming | 2006-10-04 20:42:06 -0500 (Wed, 04 Oct 2006) - | 2 lines this change was correct, the old version is no longer - needed ........ - -2006-10-05 01:40 +0000 [r44435] Steve Murphy <murf@digium.com> - - * main/pbx.c, apps/app_read.c, apps/app_waitforring.c, CHANGES, - apps/app_speech_utils.c: As per ToDo list, I have made it so that - Wait(), WaitExten(), Congestion(), Busy(), Read(), WaitForRing(), - will now either actually handle a floating point argument as - advertised, or has been upgraded to accept a floating point - [timeout] arg. - -2006-10-05 01:30 +0000 [r44434] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44433 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44433 | kpfleming | 2006-10-04 20:30:05 -0500 - (Wed, 04 Oct 2006) | 10 lines Merged revisions 44432 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04 - Oct 2006) | 2 lines fix Polycom presence notification again - ........ ................ - -2006-10-04 23:52 +0000 [r44408-44423] Luigi Rizzo <rizzo@icir.org> - - * configure.ac: simplify checks for OSS using AST_EXT_LIB_CHECK; - remove two repeated blocks using better logic. - - * acinclude.m4: small formatting fix - - * acinclude.m4: when only checking headers, do not set $1_LIB. Also - PBX_$1=0 is the default so we don't need to set it explicitly. - - * acinclude.m4: document, and extend a bit the macro - AST_EXT_LIB_CHECK so that it can be used in more places in - configure.ac - - * configure.ac: restore proper CPPFLAGS and LDFLAGS for FreeBSD, - until a better solution is found. Please do not commit the - regenerated "configure" file yet, as there are some more - simplifications to be applied to configure.ac and acinclude.m4 in - the next few days. For the same reason, i am postponing the - commit to the 1.4 branch until the above changes are complete. - - * utils/Makefile: correct libraries for astman, at least so i - think... - - * Makefile: put linker flags in ASTLDFLAGS where they belong - -2006-10-04 21:20 +0000 [r44379-44394] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44393 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44393 | kpfleming | 2006-10-04 16:17:30 -0500 - (Wed, 04 Oct 2006) | 11 lines Merged revisions 44392 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04 - Oct 2006) | 3 lines remove workaround for old Polycom firmware - SUBSCRIBE requests add workaround for new Polycom firmware - SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware) - ........ ................ - - * include/asterisk.h, /, main/utils.c: Merged revisions 44390 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44390 | kpfleming | 2006-10-04 16:04:21 -0500 (Wed, 04 Oct 2006) - | 2 lines make LOW_MEMORY builds actually work ........ - - * include/asterisk/utils.h, main/autoservice.c, main/dnsmgr.c, - channels/chan_zap.c, res/res_snmp.c, /, apps/app_meetme.c, - channels/chan_sip.c, main/utils.c, main/devicestate.c, - res/res_musiconhold.c, res/res_jabber.c, apps/app_queue.c, - channels/chan_iax2.c, channels/chan_oss.c, main/cdr.c, - channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c, - res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c, - main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c, - apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: - Merged revisions 44378 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) - | 4 lines update thread creation code a bit reduce standard - thread stack size slightly to allow the pthreads library to - allocate the stack+data and not overflow a power-of-2 allocation - in the kernel and waste memory/address space add a new stack size - for 'background' threads (those that don't handle PBX calls) when - LOW_MEMORY is defined ........ - -2006-10-04 19:33 +0000 [r44336-44377] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.tab.c, - pbx/ael/ael.y, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test16 - (added), pbx/ael/ael-test/ael-test16/extensions.ael: These - changes resolve the problems in bug 8090, where there's a crash - compiling an empty context - - * configs/muted.conf.sample: I've been meaning to add some - explanation about muted... here it is - - * configs/manager.conf.sample: CLI reverbification update to this - config file - - * apps/app_macro.c: Added a warning to the documentation for Macro - in response to bug 7776 - -2006-10-04 00:26 +0000 [r44323] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, include/asterisk.h, /, main/asterisk.c, main/loader.c, - main/term.c: Merged revisions 44322 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44322 | kpfleming | 2006-10-03 19:25:44 -0500 (Tue, 03 Oct 2006) - | 3 lines ensure that local include files are always used avoid a - duplicate function name (term_init()) ........ - -2006-10-03 22:36 +0000 [r44313] Matt O'Gorman <mogorman@digium.com> - - * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions - 44312 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44312 - | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2 - lines fix issue with dialing client without resource. ........ - -2006-10-03 20:19 +0000 [r44299] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_queue.c: Merged revisions 44298 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44298 | kpfleming | 2006-10-03 15:18:29 -0500 - (Tue, 03 Oct 2006) | 10 lines Merged revisions 44296 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44296 | kpfleming | 2006-10-03 15:14:13 -0500 (Tue, 03 - Oct 2006) | 2 lines fix a logic error in my previous fix to the - queue reload code ........ ................ - -2006-10-03 20:17 +0000 [r44297] Joshua Colp <jcolp@digium.com> - - * CHANGES, apps/app_queue.c: Strat becomes Strategy based on - feedback from two nameless fellows - -2006-10-03 18:47 +0000 [r44287] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx: Merged revisions 44283,44286 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44283 | pcadach | 2006-10-04 00:30:48 +0600 (Срд, 04 Окт 2006) | - 1 line Fix preparation of type and presentation of calling number - ........ r44286 | pcadach | 2006-10-04 00:42:20 +0600 (Срд, 04 - Окт 2006) | 1 line Change default presentation indicator to "user - provided not screened" if octet 3a missed in CallingPartyNumber - IE ........ - -2006-10-03 18:37 +0000 [r44273-44285] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 44284 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44284 | file | 2006-10-03 14:35:55 -0400 (Tue, 03 Oct 2006) | 2 - lines Use VideoSupport instead so it is considered a valid XML - attribute name. (issue #8075 reported by renemendoza) ........ - - * CHANGES, apps/app_queue.c: Add 'Strat' manager field to - QueueParams event. (issue #7704 reported by renemendoza) - - * main/channel.c, CHANGES: Add Masquerade manager event which trips - when a masquerade happens (issue #7840 reported by moy) - -2006-10-03 16:42 +0000 [r44263] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, - pbx/ael/ael-test/ref.ael-ntest10, - pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, - pbx/ael/ael-test/ref.ael-vtest13: These changes correspond to the - changes to app_stack's Gosub() application - -2006-10-03 16:15 +0000 [r44262] Joshua Colp <jcolp@digium.com> - - * UPGRADE.txt: First entry! Tell people about the callerid changes - with manager. - -2006-10-03 15:53 +0000 [r44253] Matt O'Gorman <mogorman@digium.com> - - * main/udptl.c, funcs/func_rand.c, main/say.c, apps/app_record.c, - apps/app_test.c, funcs/func_strings.c, apps/app_alarmreceiver.c, - apps/app_ices.c, channels/chan_iax2.c, main/loader.c, - res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c, - apps/app_zapras.c, main/http.c, channels/chan_alsa.c, - apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, main/sched.c, - apps/app_dial.c, main/pbx.c, channels/chan_agent.c, - apps/app_disa.c, channels/iax2-provision.c, - apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c, - channels/chan_misdn.c, apps/app_zapbarge.c, - channels/chan_features.c, apps/app_macro.c, apps/app_voicemail.c, - apps/app_meetme.c, res/res_musiconhold.c, channels/chan_gtalk.c, - res/res_jabber.c, main/enum.c, cdr/cdr_csv.c, main/channel.c, - channels/chan_phone.c, apps/app_osplookup.c, main/manager.c, - apps/app_mp3.c, res/res_agi.c, main/logger.c, main/app.c, - main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c, - res/res_config_pgsql.c, channels/chan_zap.c, funcs/func_db.c, - channels/chan_sip.c, apps/app_festival.c, - apps/app_waitforsilence.c, res/res_adsi.c, res/res_crypto.c, - apps/app_queue.c, main/rtp.c, cdr/cdr_tds.c, - channels/chan_jingle.c, apps/app_directed_pickup.c, main/file.c, - pbx/pbx_dundi.c, channels/chan_nbs.c, main/dsp.c: bug #8076 check - option_debug before printing to debug channel. patch provided in - bugnote, with minor changes. - -2006-10-03 15:50 +0000 [r44252] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Okay, I can't use ast_app_separate_args for - that... and add some debugging for murf... - -2006-10-03 15:48 +0000 [r44250-44251] Luigi Rizzo <rizzo@icir.org> - - * configure.ac: comment the fact that autoconf2.59 is ok to process - this file, but we want to use 2.60 in case the generated - "configure" file must me committed back to the repository, so we - keep differences to a minimum. - - * bootstrap.sh: simplify this file - -2006-10-03 00:07 +0000 [r44241] Matt O'Gorman <mogorman@digium.com> - - * include/asterisk/jabber.h, res/res_jabber.c: 44240 same as but - without the removing of chan_jingle and such, as I hope to finish - jingle support for 1.6 - -2006-10-02 22:02 +0000 [r44231] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Use the standard parsing routines - -2006-10-02 20:58 +0000 [r44200-44218] Joshua Colp <jcolp@digium.com> - - * configs/queues.conf.sample, doc/channelvariables.txt, CHANGES, - apps/app_queue.c: Expand setinterfacevar option to also set a - variable, MEMBERNAME, which contains the member's name. (issue - #8046 reported by jmls) - - * apps/app_dial.c, main/channel.c, apps/app_meetme.c, - res/res_features.c, apps/app_dumpchan.c, CHANGES, - apps/app_queue.c: Make callerid fields in Manager events more - consistent. CallerIDNum for number and CallerIDName for name. - (issue #7976 reported by suhler) - - * /, channels/chan_sip.c: Merged revisions 44215 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44215 | file | 2006-10-02 16:11:02 -0400 (Mon, - 02 Oct 2006) | 10 lines Merged revisions 44213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2 - lines Change the fd on the I/O context in case it changed during - the reload, which is indeed possible. (issue #7943 reported by - eclubb) ........ ................ - - * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 44199 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44199 | file | 2006-10-02 15:41:39 -0400 (Mon, - 02 Oct 2006) | 10 lines Merged revisions 44198 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r44198 | file | 2006-10-02 15:39:59 -0400 (Mon, 02 Oct 2006) | 2 - lines We should be using $AST_SBIN instead of hardcoding the path - for the error message (issue #7942 reported by eclubb) ........ - ................ - -2006-10-02 19:01 +0000 [r44187-44196] Paul Cadach <paul@odt.east.telecom.kz> - - * configs/users.conf.sample, /, pbx/pbx_config.c: Merged revisions - 44186 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) | - 1 line Missed part of userconf functionality for chan_h323 - ........ - - * /, doc/realtime.txt: Merged revisions 44167 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44167 | pcadach | 2006-10-02 23:16:37 +0600 (Пнд, 02 Окт 2006) | - 1 line Typo fix ........ - - * /, channels/chan_h323.c: Merged revisions 44166 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44166 | pcadach | 2006-10-02 23:15:11 +0600 (Пнд, 02 Окт 2006) | - 1 line Optimization of oh323_indicate(): less locks - less - problems, plus single exit point ........ - -2006-10-02 17:54 +0000 [r44153-44172] Joshua Colp <jcolp@digium.com> - - * main/logger.c, CHANGES, configs/logger.conf.sample: Add option to - logger to rename log files with timestamp (issue #8020 reported - by jmls) - - * /, main/io.c: Merged revisions 44169 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44169 | file | 2006-10-02 13:25:13 -0400 (Mon, - 02 Oct 2006) | 10 lines Merged revisions 44168 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r44168 | file | 2006-10-02 13:22:27 -0400 (Mon, 02 Oct 2006) | 2 - lines Shrink when current_ioc is unused. It is set to -1 when - unused, not 0. (issue #7941 reported by eclubb) ........ - ................ - - * res/res_monitor.c: Get rid of the IS_NULL_STRING macro and use - ast_strlen_zero instead (issue #8070 reported by wrmem) - -2006-10-02 16:00 +0000 [r44152] Kevin P. Fleming <kpfleming@digium.com> - - * main/asterisk.c: clean up formatting and conformance to code - guidelines revert Mark's change that caused a memory leak - (cap_set_proc() does not free the capability structure so we - always need to call cap_free()) - -2006-10-02 15:40 +0000 [r44150] Joshua Colp <jcolp@digium.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option - 'keepstats' which will keep queue statistics during a reload. - (issue #7908 reported by jmls) - -2006-10-02 04:17 +0000 [r44148] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: It makes more sense that in GosubIf that the - two labels might have different arguments. - -2006-10-02 02:38 +0000 [r44145-44147] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Don't use channel when - you don't mean a channel - - * main/asterisk.c: Uhm yah, not sure who committed this into - trunk... Anyway, I think this is what was intended... - -2006-10-01 19:40 +0000 [r44136] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/chan_h323.c: Merged revisions 44135 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44135 | pcadach | 2006-10-02 01:32:24 +0600 (Пнд, 02 Окт 2006) | - 1 line Do not simulate any audio tones if we got PROGRESS message - ........ - -2006-10-01 18:30 +0000 [r44112-44126] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 44125 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44125 | russell | 2006-10-01 14:30:06 -0400 (Sun, 01 Oct 2006) | - 6 lines Fix a problem that cuased AST_DATA_DIR in defaults.h to - be empty. The cause is that since ASTDATADIR is explicitly - exported using "export ASTDATADIR" at the top of the Makefile, - make no longer considers the variable "undefined", so the - Makefile can't use ?= to set ASTDATADIR if not yet set. (issue - #8063, reported by akohlsmith, fixed by me) ........ - - * /, configs/queues.conf.sample: Merged revisions 44111 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r44111 | russell | 2006-10-01 11:20:12 -0400 - (Sun, 01 Oct 2006) | 11 lines Merged revisions 44110 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 - Oct 2006) | 3 lines Fix the name of the "eventmemberstatus" - option in the sample queues.conf (issue #8065, adamg) ........ - ................ - -2006-10-01 05:37 +0000 [r44100] Tilghman Lesher <tlesher@digium.com> - - * apps/app_zapateller.c: Janitor for Zapateller: convert to use - argument macros - -2006-09-30 19:23 +0000 [r44091] Paul Cadach <paul@odt.east.telecom.kz> - - * /, main/rtp.c: Merged revisions 44090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) | - 1 line Allow one-way RTP streams (device->Asterisk) ........ - -2006-09-30 16:37 +0000 [r44081] Luigi Rizzo <rizzo@icir.org> - - * Makefile, main/Makefile, codecs/lpc10/Makefile: merge compile - fixes from 44080: - with AST_DEVMODE, building codecs/lpc10 fails - because of lots of warnings, and the configure step in editline - fails as well. Fix this by removing the -Werror in these steps. - - on FreeBSD (but probably on other platforms as well), the final - link of asterisk fails because AST_LIBS was not exported to the - subdirs Makefiles. Add a proper fix in the top-level Makefile (a - possible alternative way is to add "export AST_LIBS" near the - beginning of the file). With this fix, i believe that some of the - platform-specific conditionals in main/Makefile are redundant - (because they should be already dealt with in the top level - Makefile) but i don't have a platform to check. - -2006-09-30 16:15 +0000 [r44069-44079] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/chan_sip.c: Merged revisions 44078 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | - 6 lines Fix issue #7928 correctly. Next is a comment of previous - fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan - with a small fix by me, myself or I. Thanks, Philippe! (This was - caused by my changes to the transaction handling) ........ - - * /, channels/chan_sip.c: Merged revisions 44068 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | - 14 lines Found some buggy SIP clients (phones Planet VIP-153T - firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK - not on OK message only (when remote party answers) but on RINGING - message too, so when we send 200 OK message, we get unidentified - ACK message (because INVITE acknowledged on RINGING message - already), so 200 OK retransmits within its retransmission - interval then call gets dropped. If someone else knows how to - provide workaround for such cases, please, fix it in correct way. - Thanks to ssh from #asteriskru for provide access to his box to - study and fix this case. ........ - -2006-09-29 22:52 +0000 [r44056-44058] Kevin P. Fleming <kpfleming@digium.com> - - * /, agi, utils: Merged revisions 44057 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44057 | kpfleming | 2006-09-29 17:51:53 -0500 (Fri, 29 Sep 2006) - | 2 lines ignore temporary files made by the Makefiles during a - build ........ - - * /, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, - Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile, - agi/Makefile, codecs/Makefile, utils/Makefile, configure, - build_tools/embed_modules.xml, codecs/ilbc/Makefile, - codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions - 44055 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44055 | kpfleming | 2006-09-29 17:47:40 -0500 (Fri, 29 Sep 2006) - | 2 lines fix a few build system bugs, and convert Makefiles to - be compatible with GNU make 3.80 ........ - -2006-09-29 22:36 +0000 [r44054] Jason Parker <jparker@digium.com> - - * /, main/asterisk.c, main/cli.c: Merged revisions 44053 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44053 | qwell | 2006-09-29 15:35:09 -0700 (Fri, 29 Sep 2006) | 3 - lines Fix a bug with the removal of 'atleast' argument to 'core - verbose' and 'core debug'. Add that argument back in. ........ - -2006-09-29 21:13 +0000 [r44044] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx: Merged revisions 44034,44042-44043 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | - 1 line Fake display name by called number on incoming calls - (until passing connected number/connected name is not - implemented) ........ r44042 | pcadach | 2006-09-30 03:05:43 - +0600 (Сбт, 30 Сен 2006) | 1 line Set TON/PRESENTATION - information more carefully when no CallingNumber IE available - ........ r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 - Сен 2006) | 1 line Compile first, please ........ - -2006-09-29 20:16 +0000 [r44033] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Remove locking conflict - -2006-09-29 19:16 +0000 [r44024-44025] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Merged - revisions 44022 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44022 | pcadach | 2006-09-30 01:06:55 +0600 (Сбт, 30 Сен 2006) | - 3 lines Properly pass TON/PRESENTATION information - original - H323Connection::SendSignalSetup() destroys Q.931 fields. ........ - -2006-09-29 18:54 +0000 [r44013] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, codecs/Makefile, utils/Makefile, /, configure, - include/asterisk/autoconfig.h.in, main/Makefile, - codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, - Makefile.rules, pbx/Makefile, channels/Makefile, - main/db1-ast/Makefile: Merged revisions - 43996-43997,44008,44011-44012 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006) - | 2 lines another cross-compile fix ........ r43997 | kpfleming | - 2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines support - --without-curl in configure script ........ r44008 | kpfleming | - 2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines don't - abuse CFLAGS and LDFLAGS for build of Asterisk components, - because they are also then used for non-Asterisk components (like - menuselect); use our own variables instead ........ r44011 | - kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2 - lines missed one conversion to ASTCFLAGS ........ r44012 | - kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2 - lines yet another place where we were not using the correct - CFLAGS by default ........ - -2006-09-29 18:35 +0000 [r44010] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged - revisions 44009 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r44009 | pcadach | 2006-09-30 00:30:34 +0600 (Сбт, 30 Сен 2006) | - 1 line Pass TON/PRESENTATION information too ........ - -2006-09-29 16:38 +0000 [r43979-43994] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 43993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43993 | kpfleming | 2006-09-29 11:38:27 -0500 (Fri, 29 Sep 2006) - | 2 lines a couple more environment settings that can't leak into - the menuselect build ........ - - * /, main/cli.c: Merged revisions 43978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43978 | kpfleming | 2006-09-29 08:45:40 -0500 (Fri, 29 Sep 2006) - | 2 lines proper fix for ast_group_t change ........ - -2006-09-29 01:36 +0000 [r43954-43961] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: One must remember to unlock their list... - thanks to Qwell for letting me into his box - - * main/pbx.c: Cache the application pointer so we don't have to - needlessly search for it over and over. This should yield a - suitable performance increase. - -2006-09-28 22:43 +0000 [r43953] Kevin P. Fleming <kpfleming@digium.com> - - * /, include/asterisk/lock.h: Merged revisions 43952 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43952 | kpfleming | 2006-09-28 17:42:18 -0500 (Thu, 28 - Sep 2006) | 2 lines eliminate compiler warning when - DEBUG_CHANNEL_LOCKS is enabled and users of this header file - don't also include channel.h ........ - -2006-09-28 20:13 +0000 [r43945] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 43944 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43944 | qwell | 2006-09-28 13:11:22 -0700 (Thu, 28 Sep 2006) | 4 - lines Fix incorrect argument order for member names, on persisted - members. Issue 8047, patch by jmls. ........ - -2006-09-28 18:09 +0000 [r43934] Joshua Colp <jcolp@digium.com> - - * main/udptl.c, main/frame.c, /, channels/chan_sip.c, - funcs/func_timeout.c, apps/app_festival.c, - apps/app_alarmreceiver.c, channels/iax2-provision.c, - res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c, - res/res_monitor.c, apps/app_playback.c, - include/asterisk/logger.h, res/res_smdi.c, channels/chan_misdn.c, - channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c: - Merged revisions 43933 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 - lines Put in missing \ns on the end of ast_logs (issue #7936 - reported by wojtekka) ........ - -2006-09-28 17:38 +0000 [r43921] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_queue.c: Merged revisions 43919 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43919 | kpfleming | 2006-09-28 12:35:42 -0500 - (Thu, 28 Sep 2006) | 10 lines Merged revisions 43916 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28 - Sep 2006) | 2 lines fix buggy (and overly complex) loop used - during reload of app_queue for static member list updating - ........ ................ - -2006-09-28 17:36 +0000 [r43920] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx: Merged revisions 43918 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43918 | pcadach | 2006-09-28 23:34:19 +0600 (Чтв, 28 Сен 2006) | - 1 line Extend call establishment timeout ........ - -2006-09-28 17:32 +0000 [r43912-43917] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 43915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2 - lines Make sure the pvt exists before accessing it again as it - may have gone away (issue #7562 reported by Seb7 and issue #7939 - reported by sorg) ........ - - * /, main/cli.c: Merged revisions 43913 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43913 | file | 2006-09-28 13:14:07 -0400 (Thu, 28 Sep 2006) | 2 - lines Warning be gone! ........ - - * channels/chan_sip.c: Add jitterbuffer information to sip list - settings (issue #7945 reported by sergee) - -2006-09-28 16:54 +0000 [r43902] BJ Weschke <bweschke@btwtech.com> - - * /, apps/app_queue.c: Merged revisions 43899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43899 | bweschke | 2006-09-28 12:41:05 -0400 - (Thu, 28 Sep 2006) | 11 lines Merged revisions 43897 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 - Sep 2006) | 3 lines app_queue is comparing the device names - incorrectly while checking their statuses. It's internal list of - interfaces includes the dial string, while the argument passed to - this function does not have the dial string (/n for a local - channel). This causes it to ignore the device state changes - because it thinks it belongs to none of its members. (#8040 - reported and patch by tim_ringenbach) ........ ................ - -2006-09-28 16:43 +0000 [r43900] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/cli.c: Merged revisions 43898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43898 | kpfleming | 2006-09-28 11:38:25 -0500 - (Thu, 28 Sep 2006) | 10 lines Merged revisions 43895 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28 - Sep 2006) | 2 lines eliminate compiler warning introduced by - recent changes ........ ................ - -2006-09-28 16:19 +0000 [r43894] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 43893 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43893 | file | 2006-09-28 12:17:36 -0400 (Thu, - 28 Sep 2006) | 10 lines Merged revisions 43891 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2 - lines Stop the stream after waitstream returns so that our - formats get restored. (issue #7370 reported by kryptolus) - ........ ................ - -2006-09-28 16:01 +0000 [r43888] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx: Merged revisions 43877 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43877 | pcadach | 2006-09-28 21:56:21 +0600 (Чтв, 28 Сен 2006) | - 1 line Fix compiler warning ........ - -2006-09-28 15:32 +0000 [r43865-43875] BJ Weschke <bweschke@btwtech.com> - - * /, apps/app_queue.c: Merged revisions 43873 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43873 | bweschke | 2006-09-28 11:29:21 -0400 - (Thu, 28 Sep 2006) | 11 lines Merged revisions 43871 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28 - Sep 2006) | 3 lines Fix race condion crash with get_member_status - (#7864 - tim_ringenbach reported and patched) ........ - ................ - - * /, apps/app_queue.c: Merged revisions 43864 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43864 | bweschke | 2006-09-28 09:24:10 -0400 (Thu, 28 Sep 2006) - | 3 lines Autopause not working for queue members. (#8042 - jmls - reported and patch) ........ - -2006-09-28 13:02 +0000 [r43863] Paul Cadach <paul@odt.east.telecom.kz> - - * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h, - include/asterisk/compiler.h: Merged revisions 43861-43862 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | - 1 line Put attribute tag at correct place ........ r43862 | - pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line - Force remote side to start media on outgoing PROGRESS message - ........ - -2006-09-28 11:32 +0000 [r43854-43855] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/isdn_lib.c: Merged revisions 43852 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43852 | crichter | 2006-09-28 13:03:05 +0200 - (Do, 28 Sep 2006) | 9 lines Merged revisions 43764 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 - Sep 2006) | 1 line fixed a bug which led to chan_list zombies, - when the call could not be properly established in misdn_call. - also removed the ACK_HDLC stuff which is not really needed. - ........ ................ - - * channels/chan_misdn.c, /, channels/Makefile: Merged revisions - 43775 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) | - 1 line removed the chan_misdn versioning, since asterisk has it's - own ........ - -2006-09-28 11:12 +0000 [r43845-43853] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/cisco-h225.h, /, channels/h323/ast_h323.cxx, - main/file.c, channels/h323/cisco-h225.asn, - channels/h323/cisco-h225.cxx: Merged revisions - 43635,43843-43844,43846 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | - 1 line Fix ASN1 description of non-standard Cisco extensions - ........ r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 - Сен 2006) | 1 line Don't treat unknown control frames as voice - ........ r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 - Сен 2006) | 1 line Don't warn on HOLD/UNHOLD control frames - ........ r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 - Сен 2006) | 1 line Do not open transmit channel until TCS is - received ........ - - * channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, CHANGES, channels/h323/chan_h323.h, - configs/h323.conf.sample: Handle HOLD/RETRIEVE notifications - -2006-09-27 22:01 +0000 [r43827-43836] Joshua Colp <jcolp@digium.com> - - * CHANGES: Update CHANGES to reflect libcap capability that was - added. - - * configure, main/Makefile, configure.ac, makeopts.in, - doc/security.txt, main/asterisk.c: Add ability to set high ToS - bits as non-root on Linux using libcap (issue #7047 reported by - maddison) - - * apps/app_voicemail.c: Finish up last commit - - * apps/app_voicemail.c: Do the directory walk dance instead of - repeated stat calls as it seems to be faster (issue #7507 - reported by Corydon76) - -2006-09-27 20:27 +0000 [r43817] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 43816 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43816 | tilghman | 2006-09-27 15:21:54 -0500 - (Wed, 27 Sep 2006) | 10 lines Merged revisions 43815 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27 - Sep 2006) | 2 lines Avoid inability to lock directory log message - by creating the directory ahead of time. (Issue 7631) ........ - ................ - -2006-09-27 20:03 +0000 [r43804-43814] Jason Parker <jparker@digium.com> - - * main/pbx.c: Add BACKGROUNDSTATUS to Background() Issue #7835, - original patch by bcnit - redone by me. - - * main/pbx.c, /, apps/app_playback.c: Merged revisions 43803 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 - lines Fix an issue with PLAYBACKSTATUS not being set under - certain circumstances. Fix a minor issue, to make it use the - filenames that were parsed, instead of the entire argument - string. Fix Background() to return -1 like Playback(), if no args - are specified. ........ - -2006-09-27 19:39 +0000 [r43792-43802] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: I *think* this is the last list in - chan_iax2 - - * /, main/rtp.c: Merged revisions 43798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2 - lines Compensate for out of order packets better if RFC2833 - compensation is turned on. ........ - - * /, channels/chan_iax2.c: Merged revisions 43783 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 - lines Get rid of two functions from a time now past (we THINK - these are from pre-recursive lock time) that may be contributing - to two open issues on the bug tracker (7562/7939) and that has - the potential to just make bad things happen if the timing is - right. ........ - -2006-09-27 17:00 +0000 [r43785] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Fix some little things - -2006-09-27 16:57 +0000 [r43780] Russell Bryant <russell@digium.com> - - * main/channel.c, /, res/res_features.c: Merged revisions 43779 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43779 | russell | 2006-09-27 12:55:49 -0400 - (Wed, 27 Sep 2006) | 50 lines Merged revisions 43778 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 - Sep 2006) | 42 lines Fix a problem that occurred if a user - entered a digit that matched a bridge feature that was configured - using multiple digits, and the digit that was pressed timed out - in the feature digit timeout period. For example, if blind - transfer is configured as '##', and a user presses just '#'. In - this situation, the call would lock up and no longer pass any - frames. (issue #7977 reported by festr, and issue #7982 reported - by michaels and valuable input provided by mneuhauser and kuj. - Fixed by me, with testing help and peer review from Joshua Colp). - There are a couple of issues involved in this fix: 1) When - ast_generic_bridge determines that there has been a timeout, it - returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets - this result, it calls ast_generic_bridge over again with the same - timestamp for the next event. This results in an endless loop of - nothing until the call is terminated. This is resolved by simply - changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it - sees a timeout. 2) I also changed ast_channel_bridge such that if - in the process of calculating the time until the next event, it - knows a timeout has already occured, to immediately return - AST_BRIDGE_COMPLETE instead of attempting to bridge the channels - anyway. 3) In the process of testing the previous two changes, I - ran into a problem in res_features where ast_channel_bridge would - return because it determined that there was a timeout. However, - ast_bridge_call in res_features would then determine by its own - calculation that there was still 1 ms before the timeout really - occurs. It would then proceed, and since the bridge broke out and - did *not* return a frame, it interpreted this as the call was - over and hung up the channels. The reason for this was because - ast_bridge_call in res_features and ast_channel_bridge in - channel.c were using different times for their calculations. - channel.c uses the start_time on the bridge config, which is the - time that the feature digit was recieved. However, res_features - had another time, 'start', which was set right before calling - ast_channel_bridge. 'start' will always be slightly after - start_time in the bridge config, and sometimes enough to round up - to one ms. This is fixed by making ast_bridge_call use the same - time as ast_channel_bridge for the timeout calculation. ........ - ................ - -2006-09-27 16:49 +0000 [r43777] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c: Add CLI block and unblock circuit commands - for SS7. - -2006-09-27 16:25 +0000 [r43776] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 43774 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2 - lines Make rfc2833compensate a global option. ........ - -2006-09-27 12:32 +0000 [r43763] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Use ast_strdupa() instead of strdup(), - thanks to sergee - -2006-09-27 04:37 +0000 [r43754-43757] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: remote an unused buffer in mm_login() - (issue #8038, selsky) In passing, I have cleaned up some - formatting to better comply with our guidelines. I have also - changed one place to use S_OR(), and a couple of places to use - ast_strlen_zero() as appropriate. - -2006-09-27 03:45 +0000 [r43740-43747] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ref.ael-test5, - pbx/ael/ael-test/ael-test11/extensions.ael, - pbx/ael/ael-test/ref.ael-test6, CHANGES, - pbx/ael/ael-test/ael-test3/extensions.ael, - pbx/ael/ael-test/ref.ael-test7, - pbx/ael/ael-test/ael-test5/extensions.ael, - pbx/ael/ael-test/ref.ael-vtest13: This commits the changes to AEL - to use the gosub-with-args from Tilghman to perform macro calls. - This results in substantially smaller stack footprint, which - allows macro call depths in excess of 100,000 levels, rather than - the limit of 7 calls deep, which the Macro app is subject to. - - * /, configs/extensions.ael.sample: Merged revisions 43739 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1 - line This change to extensions.ael was to fix bug 8031; the - install scripts are causing it to be copied to - /etc/asterisk/extensions.ael, and because it is a fairly direct - conversion of the original extensions.conf, the macro and context - names clash with the existing extensions.conf. So, I put an ael- - in front of all macros and contexts, and checked every goto and - macro call. Also, this file compiles under aelparse. ........ - -2006-09-27 01:39 +0000 [r43733] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Clean up code and convert last two things - (firmware/dialplan cache) to linked list macros. - -2006-09-26 22:18 +0000 [r43721-43727] Jason Parker <jparker@digium.com> - - * apps/app_meetme.c: Fire a manager event when a meetme is - started/stopped. Issue #7891, patch by suhler. - - * apps/app_queue.c: Add QueueSummary manager action. Gives "at a - glance" information about a single queue, or all queues. Issue - #8035, patch by rgollent, slightly modified (formatting) by me. - -2006-09-26 21:01 +0000 [r43715] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: Merged revisions 43710 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43710 | russell | 2006-09-26 16:56:42 -0400 - (Tue, 26 Sep 2006) | 17 lines (This was actually BE-65) Merged - revisions 43708 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | - 7 lines Back in revision 4798, this message was changed from - using ast_cli() to directly calling write(). During this change, - checking if this was a remote console was removed. This caused - this message about using "exit" or "quit" to exit an Asterisk - console to come up in times where it did not make sense. This - change restores the check to see if this is a remote console - before printing the message. (fixes BE-4) ........ - ................ - -2006-09-26 20:51 +0000 [r43709] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c, include/asterisk/channel.h, .cleancount, - main/cli.c: Merged revisions 43707 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43707 | file | 2006-09-26 16:47:26 -0400 (Tue, - 26 Sep 2006) | 10 lines Merged revisions 43705 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2 - lines Use proper type to represent the group variable (issue - #8025 reported by makoto) ........ ................ - -2006-09-26 20:30 +0000 [r43702] Jason Parker <jparker@digium.com> - - * CHANGES: update CHANGES file to reflect codec support in - chan_skinny - -2006-09-26 20:26 +0000 [r43701] Russell Bryant <russell@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 43700 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43700 | russell | 2006-09-26 16:24:39 -0400 - (Tue, 26 Sep 2006) | 14 lines Merged revisions 43699 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26 - Sep 2006) | 6 lines When parsing the sections of voicemail.conf - that contain mailbox definitions, don't introduce a length limit - on the definition by using a 256 byte temporary storage buffer. - Instead, make the temporary buffer just as big as it needs to be - to hold the entire mailbox definition. (fixes BE-68) ........ - ................ - -2006-09-26 20:20 +0000 [r43696-43698] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 43697 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep - 2006) | 2 lines Strip options off the argument passed for - devicestate in chan_local. (issue #8034 reported by pcardozo) - ........ - - * main/channel.c, /, main/slinfactory.c, apps/app_chanspy.c: Merged - revisions 43695 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 - lines Slight overhaul of the whisper support. 1. We need to - duplicate the frame from ast_translate 2. We need to ensure we - always have signed linear coming in for signed linear combining. - 3. We need to ensure we are always feeding signed linear out. 4. - Properly store and restore write format when beeping on the - channel we are whispering on. 5. Properly discontinue the stream - on the channel for the beep. (issue #8019 reported by - timkelly1980) ........ - -2006-09-26 19:37 +0000 [r43677-43687] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: start a CHANGES file for trunk... no need to force - people to have to review commit logs after branching - - * /, sounds/Makefile: Merged revisions 43676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43676 | kpfleming | 2006-09-26 13:34:27 -0500 (Tue, 26 Sep 2006) - | 2 lines update to use 1.4.3 core sounds, with corrected - beep/beeperr/tt-monkeys files ........ - -2006-09-26 18:10 +0000 [r43675] Jason Parker <jparker@digium.com> - - * main/frame.c, /, doc/rtp-packetization.txt: Merged revisions - 43674 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43674 | qwell | 2006-09-26 11:08:51 -0700 (Tue, 26 Sep 2006) | 4 - lines Issue #8015, patch by Dan Austin. Maximum values were - incorrect, which is why this is being put in 1.4 ........ - -2006-09-26 17:25 +0000 [r43667] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c: Gosub arguments (Issue 7780) - -2006-09-26 17:09 +0000 [r43666] Jason Parker <jparker@digium.com> - - * main/logger.c, configs/logger.conf.sample: Add optional - queue_log_name config option for logger.conf, to change the name - of the queue_log file. Issue #7363, patch by Steve Davies, - slightly modified by me. - -2006-09-26 16:56 +0000 [r43658-43659] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: MailboxExists should be a dialplan - function, not an application (Issue 7503) - - * res/res_limit.c: These three are not defined on all platforms - that we support - -2006-09-26 15:35 +0000 [r43651] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 43650 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep - 2006) | 11 lines Add proper codec support to chan_skinny. Works - with at least ulaw, alaw, and g729a. This is technically a "new - feature", but there are justifications for it. I found a bug with - the recent rtp packetization changes, which caused the media - setup to fail under certain circumstances, particularly when - using allow=all, or having no allow= statements (globally or on - the device). I could have either removed the rtp packetization - features, or I could add proper codec support (which, without, I - think most people would consider to be a bug anyways). ........ - -2006-09-25 22:09 +0000 [r43641-43643] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 43642 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43642 | tilghman | 2006-09-25 17:07:44 -0500 (Mon, 25 Sep 2006) - | 2 lines Should have moved these lines up in the merge, instead - of removing them ........ - - * /, apps/app_voicemail.c: Merged revisions 43640 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43640 | tilghman | 2006-09-25 17:04:47 -0500 - (Mon, 25 Sep 2006) | 12 lines Merged revisions 43634 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43634 | tilghman | 2006-09-25 16:14:41 -0500 (Mon, 25 - Sep 2006) | 4 lines Two bugs when forwarding voicemail (Issue - 7824): 1) delete=yes was ignored 2) maxmessages was ignored - ........ ................ - -2006-09-25 20:30 +0000 [r43627] Paul Cadach <paul@odt.east.telecom.kz> - - * /: Block revision 43626 from 1.4 tree - already here - -2006-09-25 15:24 +0000 [r43617] Jason Parker <jparker@digium.com> - - * /, sounds/Makefile: Merged revisions 43616 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43616 | qwell | 2006-09-25 08:23:31 -0700 (Mon, 25 Sep 2006) | 4 - lines One more fix for sounds installation - this time for - portability. Reported to asterisk-dev mailing list. ........ - -2006-09-25 14:49 +0000 [r43604] Steve Murphy <murf@digium.com> - - * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from - crashing if trying to play an OGG moh file. - -2006-09-25 09:03 +0000 [r43571-43597] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, - channels/chan_h323.c, channels/h323/ast_h323.h, - channels/h323/chan_h323.h, configs/h323.conf.sample: Support for - negotiation and receiption of Cisco's RTP DTMF - - * channels/h323/ast_h323.cxx: Disable fastStart if requested by - remote side - - * /: Block revision 43582 - - * channels/chan_h323.c, configs/h323.conf.sample: Specify RFC2833 - payload on dtmfmode option rather than dtmfcodec option - (deprecated) - - * channels/h323/ast_h323.cxx, channels/chan_h323.c: DTMF mode is - bitmask, not valued field - - * channels/h323/caps_h323.cxx, channels/h323/caps_h323.h: Define - Cisco RTP capability - - * channels/h323/caps_h323.cxx: Specify non-standard data - independedly on OpenH323's codec name (it can be easily changed) - - * channels/chan_h323.c, channels/h323/chan_h323.h: Define DTMF - payload types - -2006-09-24 15:01 +0000 [r43554-43565] Russell Bryant <russell@digium.com> - - * /, channels/iax2-provision.c: Merged revisions 43564 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 - Sep 2006) | 5 lines Fix a CLI command registration issue where an - erroneous message claiming that "iax2 show provisioning" was - already registered. This was because this command was registering - itself as both the command, as well as the command it is - deprecating. (issue #8022, reported by bjweeks, fixed by myself) - ........ - - * /, channels/chan_iax2.c: Merged revisions 43553 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43553 | russell | 2006-09-24 09:53:35 -0400 - (Sun, 24 Sep 2006) | 12 lines Merged revisions 43552 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 - Sep 2006) | 4 lines Check to see if the channel that is - activating the IAXPEER function is actually an IAX2 channel - before proceeding to process it to avoid crashing. (issue #8017, - reported by admott, fixed by myself) ........ ................ - -2006-09-24 12:15 +0000 [r43539-43546] Paul Cadach <paul@odt.east.telecom.kz> - - * main/rtp.c: Small Cisco's RTP DTMF update - - * channels/chan_h323.c: Avoid possible deadlock on channel - destruction - - * main/rtp.c: Correct behavior on Cisco's DTMF - -2006-09-22 23:46 +0000 [r43525-43526] Kevin P. Fleming <kpfleming@digium.com> - - * /: file forgot one :-) - - * Makefile, /: Merged revisions 43524 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43524 | kpfleming | 2006-09-22 18:44:47 -0500 (Fri, 22 Sep 2006) - | 2 lines don't output the 'build complete' message when the - target being run is already going to do an installation ........ - -2006-09-22 23:34 +0000 [r43522] Joshua Colp <jcolp@digium.com> - - * /: You see nothing... - -2006-09-22 22:13 +0000 [r43519] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 43518 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep - 2006) | 4 lines Allow chan_skinny.so to be unloaded properly. - Remove reload support, since it doesn't actually...work. ........ - -2006-09-22 21:34 +0000 [r43506-43507] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This commits a change to return - MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all - goes well for bug 8004 - - * pbx/pbx_ael.c: As per bug 8004, we now return - AST_MODULE_LOAD_DECLINE when we can't read extensions.ael - -2006-09-22 20:33 +0000 [r43495-43500] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Move from h.323 to h323 command prefix - - * channels/chan_h323.c: Fix compilation warnings - - * channels/h323/compat_h323.h: Use own factory for our - OpalMediaFormats too - - * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h: Fix our - capability's factory - -2006-09-22 17:26 +0000 [r43493] Jason Parker <jparker@digium.com> - - * /, main/cli.c: Merged revisions 43492 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43492 | qwell | 2006-09-22 10:25:05 -0700 (Fri, 22 Sep 2006) | 2 - lines Make sure we explicitly set the CLI command to not be - deprecated, if it isn't. ........ - -2006-09-22 16:43 +0000 [r43488-43490] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/Makefile: Merged revisions 43489 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43489 | kpfleming | 2006-09-22 11:42:46 -0500 (Fri, 22 Sep 2006) - | 2 lines use rebuilt extra sounds ........ - - * main/channel.c, /: Merged revisions 43486 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006) - | 2 lines all the Linux systems I have don't use '__m_count' for - this field, so I don't know where this came from... ........ - -2006-09-22 15:50 +0000 [r43483-43485] Russell Bryant <russell@digium.com> - - * channels/chan_misdn.c, /: Merged revisions 43482 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43482 | russell | 2006-09-22 11:42:44 -0400 (Fri, 22 - Sep 2006) | 3 lines return AST_MODULE_LOAD_DECLIDE if mISDN could - not be configured (issue #8006, Mithraen) ........ - -2006-09-22 14:58 +0000 [r43479-43480] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/threadstorage.h: compatibility fix: use - "attribute_XXX" instead of *__attribute__ ((XXX)) so we can - handle compiler/os dependencies in our compiler.h - - * channels/chan_sip.c: style fix: move variable declaration at the - beginning of the block. - -2006-09-22 14:04 +0000 [r43478] Russell Bryant <russell@digium.com> - - * main/frame.c, /: Merged revisions 43477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43477 | russell | 2006-09-22 10:02:58 -0400 (Fri, 22 Sep 2006) | - 3 lines Suppress a compiler warning about the use of a - potentially uninitialized variable. It couldn't actually happen, - though. ........ - -2006-09-22 04:54 +0000 [r43472] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/caps_h323.cxx: Add missing include - -2006-09-22 03:09 +0000 [r43470] Jason Parker <jparker@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 43469 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep - 2006) | 4 lines First shot at unload_module in chan_skinny.. More - to come. ........ - -2006-09-21 23:55 +0000 [r43467] Matt O'Gorman <mogorman@digium.com> - - * /, include/asterisk/jabber.h, channels/chan_gtalk.c, - res/res_jabber.c: Merged revisions 43466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006) - | 2 lines updates for better compontent support ........ - -2006-09-21 23:29 +0000 [r43463-43465] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions - 43464 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006) - | 2 lines Twould help if we actually documented how the new - features in res_odbc actually work. (Oops) ........ - - * res/res_limit.c (added): Set process limits without restarting - Asterisk - -2006-09-21 22:53 +0000 [r43461] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Oh look more changes, - but these are my own! (Clean up module load functions) - -2006-09-21 22:44 +0000 [r43460] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: Suppress compiler warnings - -2006-09-21 22:32 +0000 [r43459] Joshua Colp <jcolp@digium.com> - - * channels/chan_alsa.c: Clean up chan_alsa load module function - (issue #8000 reported by Mithraen) - -2006-09-21 22:23 +0000 [r43458] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/acl.h, doc/ip-tos.txt, channels/chan_sip.c, - doc/mp3.txt, doc/ael.txt, doc/channelvariables.txt, main/acl.c: - And some deprecated APIs and modifications to documentation - -2006-09-21 22:23 +0000 [r43455-43457] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_oss.c: Merged revisions 43456 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43456 | file | 2006-09-21 18:21:40 -0400 (Thu, 21 Sep 2006) | 2 - lines Some more clean up in the load function for chan_oss (issue - #8002 reported by Mithraen with minor mods by moi) ........ - - * /, channels/chan_mgcp.c: Merged revisions 43454 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2 - lines Clean up chan_mgcp's module load function (issue #8001 - reported by Mithraen with mods by moi) ........ - -2006-09-21 21:59 +0000 [r43452] Tilghman Lesher <tlesher@digium.com> - - * doc/ip-tos.txt, channels/chan_local.c, channels/chan_sip.c, - res/res_features.c, channels/chan_agent.c, res/res_convert.c, - res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c, - channels/chan_oss.c, channels/chan_skinny.c, - channels/chan_features.c, res/res_agi.c, channels/chan_h323.c, - channels/chan_alsa.c, apps/app_settransfercapability.c (removed), - res/res_indications.c, pbx/pbx_config.c, res/res_odbc.c, - channels/chan_mgcp.c: Lots more removal of deprecated things - -2006-09-21 21:22 +0000 [r43451] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/Makefile, build_tools/strip_nonapi (added): Merged - revisions 43450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43450 | kpfleming | 2006-09-21 16:21:29 -0500 (Thu, 21 Sep 2006) - | 2 lines add another attempt to strip non-API symbols from the - final binary... script will need to be extended to work on - non-Linux systems ........ - -2006-09-21 21:17 +0000 [r43442-43449] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, main/pbx.c, main/frame.c, main/translate.c, - apps/app_queue.c, main/config.c, main/rtp.c, - apps/app_setcdruserfield.c (removed), main/cli.c, main/channel.c, - main/manager.c, main/file.c, main/http.c, main/logger.c, - main/astmm.c, main/image.c, main/asterisk.c: Remove deprecated - CLI apps from the core - - * /, apps/app_url.c: Merged revisions 43445 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43445 | tilghman | 2006-09-21 15:22:43 -0500 (Thu, 21 Sep 2006) - | 2 lines Fix documentation to reflect how Url() really works - ........ - - * apps/app_setcallerid.c, apps/app_voicemail.c: More removal of - deprecated stuff - - * main/pbx.c, main/manager.c, UPGRADE.txt: Remove 1.4 changes from - UPGRADE.txt, remove deprecated callerid field, remove deprecated - SetGlobalVar app - - * /, apps/app_rpt.c: Merged revisions 43441 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43441 | tilghman | 2006-09-21 14:43:32 -0500 (Thu, 21 Sep 2006) - | 2 lines Oops, missed the merge breakage ........ - -2006-09-21 19:42 +0000 [r43440] Kevin P. Fleming <kpfleming@digium.com> - - * makeopts.in: fix this so chan_zap links properly again - -2006-09-21 19:35 +0000 [r43439] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_language.c (removed), funcs/func_moh.c (removed), - apps/app_lookupcidname.c (removed), funcs/func_md5.c, - apps/app_hasnewvoicemail.c (removed), funcs/func_blacklist.c - (added), apps/app_random.c (removed), funcs/func_vmcount.c - (added), res/res_realtime.c (added), apps/app_lookupblacklist.c - (removed), apps/app_realtime.c (removed), apps/app_queue.c: - Remove deprecated apps and funcs - -2006-09-21 19:27 +0000 [r43437] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, main/channel.c, /, channels/chan_sip.c, - include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c: - SS7 marked the start of an open season for trunk again but here's - something minor - abstract early bridging into the technology so - that we don't always assume they use RTP and try it. - -2006-09-21 19:22 +0000 [r43436] Kevin P. Fleming <kpfleming@digium.com> - - * configure: regenerated at PCadach's request - -2006-09-21 19:18 +0000 [r43429-43434] Paul Cadach <paul@odt.east.telecom.kz> - - * acinclude.m4: Check for 64-bit OpenH323/PWLib versions too, - thanks to Mithraen (please, re-build configure script) - - * channels/h323/caps_h323.cxx: Declare our own media formats to not - rely on OpenH323 configuration - - * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, - channels/chan_h323.c, channels/h323/caps_h323.h: Introduce Cisco - G.726-32 capability (g726aal2 form) - -2006-09-21 18:42 +0000 [r43427-43428] Matthew Fredrickson <creslin@digium.com> - - * configure: Update configure - - * channels/chan_zap.c, build_tools/menuselect-deps.in, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, - configs/zapata.conf.sample: Merge in SS7 changes.... need to - still cleanup zapata.conf - -2006-09-21 17:06 +0000 [r43411-43423] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_rpt.c: Merged revisions 43422 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43422 | tilghman | 2006-09-21 12:04:40 -0500 - (Thu, 21 Sep 2006) | 10 lines Merged revisions 43420 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43420 | tilghman | 2006-09-21 12:01:48 -0500 (Thu, 21 - Sep 2006) | 2 lines Whitespace change... really just an excuse to - test repotools ........ ................ - - * /: Last merge should not have brought in the 1.2 props - - * /, configure, configure.ac, cdr/cdr_tds.c: Merged revisions 43410 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 - ................ r43410 | tilghman | 2006-09-21 11:31:59 -0500 - (Thu, 21 Sep 2006) | 10 lines Merged revisions 43409 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r43409 | tilghman | 2006-09-21 11:18:19 -0500 (Thu, 21 - Sep 2006) | 2 lines TDS 0.64 updates ........ ................ - -2006-09-21 16:09 +0000 [r43403-43406] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/Makefile: Merged revisions 43405 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r43405 | kpfleming | 2006-09-21 11:08:03 -0500 (Thu, 21 Sep 2006) - | 2 lines remove this change... it requires binutils 2.17 - ........ - - * /: remove extraneous property - -2006-09-20 23:20 +0000 [r43397] Jason Parker <jparker@digium.com> - - * /, build_tools/make_version: Merged revisions 43396 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r43396 | qwell | 2006-09-20 16:19:25 -0700 (Wed, 20 Sep - 2006) | 2 lines fix minor typo in the way version is handled - ........ - -2006-09-20 23:02 +0000 [r43393] Kevin P. Fleming <kpfleming@digium.com> - - * /: this has been manually merged - -2006-09-20 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta1 released. - |