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diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..7facc8993 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +36 diff --git a/.version b/.version new file mode 100644 index 000000000..c2f9575bf --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.2.0-rc2 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..1e876d5b0 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,18480 @@ +2009-09-17 Leif Madsen <lmadsen@digium.com> + + * Released Asterisk 1.6.2.0-rc2 + +2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com> + + * main/channel.c, /, include/asterisk/cdr.h, + include/asterisk/channel.h: Merged revisions 219139 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 + (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep + 2009) | 10 lines Prevent a potential race condition and crash + when hanging up a channel by removing the channel from the + channel list before begining channel tear down. This fix may + potentially cause problems with CDR backends that access the + channel a CDR is associated with via the channel list. This fix + makes the channel unavabile at the time when the CDR backend is + invoked. This has been documented in include/asterisk/cdr.h. + (closes issue #15316) Reported by: vmarrone Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/362/ ........ + ................ + +2009-09-16 23:52 +0000 [r219063] 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 19:40 +0000 [r218938] 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 19:29 +0000 [r218937] 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 19:25 +0000 [r218934] 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 13:38 +0000 [r218802] 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 22:46 +0000 [r218733] 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 19:24 +0000 [r218688] 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 16:18 +0000 [r218590] 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 16:06 +0000 [r218582] 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 15:59 +0000 [r218576] 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 ........ ................ + +2009-09-15 15:42 +0000 [r218507-218575] 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 15:05 +0000 [r218503] 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 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com> + + * sounds/Makefile, apps/app_voicemail.c, /, + configs/voicemail.conf.sample: 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 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 218295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 | + file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do + not attempt to add a parking extension if an error occurred while + reading the configuration. ........ + +2009-09-14 15:20 +0000 [r218238] 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 22:12 +0000 [r218219] 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 + +2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1 + line get rid of mfcr2 monitor thread condition, is problematic + ........ + +2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 218050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 | + tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines + Check the origination priority for more matches, not the current + priority. Found by Pavel Troller on the -dev list. ........ + + * 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) ........ + ................ + + * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /, + channels/chan_sip.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 21:21 +0000 [r217821] David Vossel <dvossel@digium.com> + + * 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 19:56 +0000 [r217739] 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 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 | + oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines + Since it's possible to have more than 999 calls, I'm changing the + call counter roof to something higher. ........ + +2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_odbc.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 217638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 | + tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines + Verify support for wide ODBC character types before using them. + (closes issue #15870) Reported by: nic_bellamy ........ + +2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1 + line ast_log replaced for ast_verbose in MFCR2 event + notifications ........ + +2009-09-10 12:09 +0000 [r217594] 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 20:37 +0000 [r217519] 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 + https://origsvn.digium.com/svn/asterisk/trunk + +2009-09-09 10:58 +0000 [r217369] 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 22:20 +0000 [r217299] Sean Bright <sean@malleable.com> + + * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | + seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 + lines Fix compilation of app_meetme. Reported by ebroad in + #asterisk-bugs ........ + +2009-09-08 20:33 +0000 [r217217] 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 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com> + + * /, configure, 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 15:36 +0000 [r217036] 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 14:27 +0000 [r216994] 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 16:43 +0000 [r216647-216845] 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 ........ + + * CHANGES, UPGRADE.txt: Add docs + + * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /, + channels/chan_sip.c, apps/app_disa.c: 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 19:42 +0000 [r216598] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | + dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines + 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 Review: + https://reviewboard.asterisk.org/r/354/ ........ + +2009-09-04 19:32 +0000 [r216597] 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 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009) + | 2 lines Fix trunk breakage. ........ + + * UPGRADE-1.6.txt, main/pbx.c, /: 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 15:11 +0000 [r216469-216509] 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 ........ ................ + + * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) + | 2 lines make sure canlog is set so we can compile with + DEBUG_THREADS enabled on OpenBSD ........ + +2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | + russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines + Do not treat every SIP peer as if they were configured with + insecure=port. There was a problem in the function responsible + for doing peer matching by IP address and port number such that + during the second pass for checking for a peer configured with + insecure=port, it would end up treating every peer as if it had + been configured that way. These changes fix the logic in the peer + IP and port comparison callback to handle insecure=port checking + properly. This problem was introduced when SIP peers were + converted to astobj2. Many thanks to dvossel for noticing this + while working on another peer matching issue. ........ + + * 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 06:14 +0000 [r216225] 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 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com> + + * /, 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 18:42 +0000 [r216007] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, 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: 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 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | + oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add + known internal IP address when autodomain=yes (closes issue + #14573) Reported by: pj Patches: sip-internip-autodomain1.diff + uploaded by mnicholson (license 96) modified by oej Tested by: pj + ........ + + * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show + channelstats". Not directly mergeable in svn trunk, needs more + tests, therefore committed directly to 1.6.2. (closes issue + #15819) Reported by: klaus3000 Patches: + asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded + by klaus3000 (license 65) Tested by: klaus3000, oej + +2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info> + + * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 | + mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines + Document that SIPshowpeer and SKINNYshowline now include the + configured parkinglot in their response. Prodded by snuff-work on + #asterisk-dev IRC channel ........ + +2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 | + tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines + Default the callback extension to "s". This is a regression. + (closes issue #15764) Reported by: elguero Change-type: bugfix + ........ + +2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) + | 25 lines Merged revisions 215682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) + | 18 lines Re-send non-100 provisional responses to prevent + cancellation From section 13.3.1.1 of RFC 3261: If the UAS + desires an extended period of time to answer the INVITE, it will + need to ask for an "extension" in order to prevent proxies from + canceling the transaction. A proxy has the option of canceling a + transaction when there is a gap of 3 minutes between responses in + a transaction. To prevent cancellation, the UAS MUST send a + non-100 provisional response at every minute, to handle the + possibility of lost provisional responses. (closes issue #11157) + Reported by: rjain Tested by: twilson Review: + https://reviewboard.asterisk.org/r/315/ ........ ................ + +2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | + dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines + port string to int conversion using sscanf There are several + instances where a port is parsed from a uri or some other source + and converted to an int value using atoi(), if for some reason + the port string is empty, then a standard port is used. This + logic is used over and over, so I created a function to handle it + in a safer way using sscanf(). ........ + +2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions + 215665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | + mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines + add Parkinglot info to sip show peer <foo> and skinny show line + <foo> If we had this from the start, debugging the 'parking not + using configured parkinglot' bug would have been easier. ........ + + * /, main/features.c: Merged revisions 215622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 | + mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines + - lock channel before looking for a channel variable - Init the + parkings list member of struct parkinglot. Thanks Sean for the + explanation why this should be here. ........ + +2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com> + + * /, main/Makefile, main/app.c: Merged revisions 215567 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02 + Sep 2009) | 9 lines Close up to the soft open file limit (same on + Linux, but varies drastically on OS X). Also, a Makefile fix for + Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: + 20090901__issue14542.diff.txt uploaded by tilghman (license 14) + Tested by: jtodd, tilghman Change-type: bugfix ........ + + * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 | + tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines + Fix register such that lines with a transport string, but without + an authuser, parse correctly. (AST-228) ........ + +2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | + dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines + SIP uri parsing cleanup Now, the scheme passed to parse_uri can + either be a single scheme, or a list of schemes ',' delimited. + This gets rid of the whole problem of having to create two + buffers and calling parse_uri twice to check for separate + schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ + +2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) + | 3 lines like in chan_sip's sip_new skinny should copy the + configured parkinglot from a line to the newly created channel. + This makes callparking honor the configured parkinglot for skinny + lines as well. ........ + +2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 | + dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines + SIP support for keep-alive event keep-alive events are used by + Sipura/Linksys for NAT keepalive. There currently don't appear to + be any problems with NAT, but everytime a keep-alive event is + received, Asterisk responds with a "489 Bad event". This error + may indicate to a user that NAT problems exist just because this + even is not supported. Now, rather than respond with an error, + the packet is consumed and a "200 ok" is sent just to indicate we + received the packet. (issue #15084) Patches: + chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) + ........ + +2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info> + + * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | + mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 + lines Honor configured parkinglot when parking and retrieving + parked calls Thank oej for pointing out the fact that sip_new did + not copy parkinglot from the peer into the newly created channel. + (closes issue #15538) Reported by: gracedman Patches: + 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak + (license 7) With mod by me to also fix callparking as well (this + uploaded patch only fixed retrieving a parked call) Tested by: + gracedman, mvanbaak ........ + +2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500 + (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) + | 12 lines Use strrchr() so SoftHangup will correctly truncate + multi-hyphen channel names In general channel names are in the + form Foo/Bar-Z, but the channel name could have multiple hyphens + and look like Foo/B-a-r-Z. Use strrchr to truncate the channel + name at the last hyphen. (closes issue #15810) Reported by: + dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard + (license 733) ........ ................ + +2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com> + + * main/frame.c, /: Merged revisions 215161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 | + kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3 + lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS + frames are properly decoded. ........ + +2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 214945 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 + (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) + | 7 lines Also unlock the "other" channel, when returning, due to + glare. (closes issue #15787) Reported by: tim_ringenbach Patches: + chan_local.diff uploaded by tim ringenbach (license 540) Tested + by: tim_ringenbach ........ ................ + + * Makefile, /: Merged revisions 214898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 | + tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines + Force Darwin on ppc platforms to compile with a target level that + supports aliasing. ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | + tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines + If lua is detected with the lua5.1 prefix (or not), adjust the + include path accordingly. Based upon feedback to a release + announcement on the -users list. See + http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html + ........ + +2009-08-29 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.0-rc1 released. + +2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 214702 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) + | 15 lines Merged revisions 214701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) + | 8 lines Modify comment to be a bit more accurate. We have kept + this comment around long enough, that it's pretty clear that + we're keeping the code, because changing the code would require a + pretty fundamental architectural shift. We've also taken + criticism in some quarters, because it was believed that it was + referring to the code being nasty. No, the code isn't nasty, just + the operation itself is rather odd. Fixed for eternity (probably + not). ........ ................ + +2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com> + + * makeopts.in, Makefile, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 214696 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | + kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 + lines Ensure that CFLAGS and/or LDFLAGS provided to configure + script are preserved. Cross-compilation environments want to + provide 'defaults' for compiler and linker options, and + frequently do this by specifying CFLAGS and LDFLAGS in the + environment or as command-line arguments to the configure script. + This patch modifies the configure script and Makefile to preserve + these settings and ensure they are used in the build process. + ........ + +2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug + 2009) | 3 lines Fix some incorrect documentation of sched_thread + functions. ........ + +2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com> + + * autoconf/libcurl.m4 (added), /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 214518 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) + | 14 lines Merged revisions 214517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) + | 7 lines Use autoconf to detect libcurl, as this enables + cross-compilation checks, something we didn't allow before. + (closes issue #15714) Reported by: pprindeville Patches: + 20090813__issue15714.diff.txt uploaded by tilghman (license 14) + Tested by: pprindeville ........ ................ + + * main/manager.c, /: Merged revisions 214514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 | + tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines + Ensure that we check for the special value + CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by: + a_villacis Patches: + asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch + uploaded by a villacis (license 660) (Plus a few of my own, to + catch the remaining places within manager.c where it could have + been a problem) ........ + + * autoconf/ast_ext_lib.m4, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 214466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) + | 9 lines Merged revisions 214436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) + | 2 lines One more build system change, to make the descriptions + look better, if we have better information. ........ + ................ + + * autoconf/ast_ext_lib.m4, /, configure, + include/asterisk/autoconfig.h.in: Merged revisions 214360 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 + (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) + | 3 lines Make autoheader descriptions render correctly in our + autoconfig.h file. (Figured out while working with issue #14906) + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | + tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines + Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue + #15362) Reported by: klaus3000 Patches: + chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license + 65) ........ + +2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com> + + * main/channel.c, /: Merged revisions 214195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) + | 25 lines Merged revisions 214194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) + | 19 lines ast_write() ignores ast_audiohook_write() results In + ast_write(), if a channel has a list of audiohooks, those lists + are written to and the resulting frame is what ast_write() should + continue with. The problem was the returned audiohook frame was + not being handled at all, and the original frame passed into it + did not contain the mixed audio, so essentially audio was being + lost. One result of this was chan_spy's whisper mode no longer + worked. To complicate the issue, frames passed into ast_write may + either be a single frame, or a list of frames. So, as the list of + frames is processed in the audiohook_write, the returned frames + had to be added to a new list. (closes issue #15660) Reported by: + corruptor Tested by: dvossel ........ ................ + +2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 214152 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | + tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines + Not all versions of gnu-linux use glibc, which contains iconv. + Some (especially embedded systems) don't have iconv at all. + (closes issue #15169) Reported by: pprindeville ........ + + * /, main/say.c: Merged revisions 214071 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009) + | 17 lines Merged revisions 214068-214069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) + | 6 lines Fix pronunciation of German dates. (closes issue + #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded + by Benjamin Kluck (license 803) ........ r214069 | tilghman | + 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should + always compile before committing... ........ ................ + + * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 | + tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines + DUNDILOOKUP function in 1.6 should use comma delimiters. (closes + issue #15322) Reported by: chappell Patches: + dundilookup-0015322.patch uploaded by chappell (license 8) + ........ + + * main/pbx.c, /: Merged revisions 213971 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009) + | 14 lines Merged revisions 213970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) + | 7 lines Improve error message by informing user exactly which + function is missing a parethesis. (closes issue #15242) Reported + by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by + dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by + loloski (license 68) ........ ................ + + * Makefile, /: Merged revisions 213904 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 | + tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines + The DTD should be installed in the same path as the rest of the + XML documentation. (closes issue #15344) Reported by: tzafrir + Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license + 46) ........ + + * Makefile, /: Merged revisions 213900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009) + | 11 lines Merged revisions 213899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009) + | 4 lines Use the default runlevels for Debian derivatives, + instead of making up our own. (closes issue #14730) Reported by: + pkempgen ........ ................ + +2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) + | 14 lines Fix storage of greetings when using IMAP_STORAGE The + store macro was not getting called preventing storage of IMAP + greetings at all. This has been corrected along with fixing + checking if the imapgreetings option is turned on to store the + greeting in IMAP. Lastly, the attachment filename was incorrectly + using the full path instead of just the basename, which was + causing problems with retrieval of the greeting. (closes issue + #14950) Reported by: noahisaac (closes issue #15729) Reported by: + lmadsen ........ + +2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1 + line improve handling of openr2_chan_disconnect_call API failure, + unlikely, but happened on openr2 library bug ........ + +2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 | + tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines + Clarifying comments in sip_register, and removing a dead section + ........ + +2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | + dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines + Register request line contains wrong address when user domain and + register host differ (closes issue #15539) Reported by: + Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by + Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel + (license 671) Tested by: Nick_Lewis, dvossel ........ + +2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug + 2009) | 12 lines Ensure that realtime mailboxes properly report + status on subscription. This patch modifies app_voicemail's + response to mailbox status subscriptions (via the internal event + system) to ensure that a subscription triggers an explicit poll + of the mailbox, so the subscriber can get an immediate cached + event with that status. Previously, the cache was only populated + with the status of non-realtime mailboxes. (closes issue #15717) + Reported by: natmlt ........ + +2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 | + dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines + fixes sip register parsing when user@domain is used (issue + #15008) (issue #15672) ........ + +2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk.h, /: Merged revisions 213560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) + | 14 lines Merged revisions 213559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) + | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. + (closes issue #15698) Reported by: slavon Patches: + 20090817__issue15698.diff.txt uploaded by tilghman (license 14) + Tested by: slavon, tilghman ........ ................ + +2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 213494 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 + (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | + 5 lines Clarify queues.conf comments to specify that variables + should be set in the dialplan. (closes issue #15755) Reported by: + trendboy ........ ................ + +2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1 + line increment the mfcr2 monitor count when clearing the call + request ........ + + * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1 + line fixed bug caused by calling ast_request without calling + ast_call on an R2 channel, ie, CHANISAVAIL ........ + +2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com> + + * main/loader.c, /: Merged revisions 213450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 | + twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines + Make LOAD_ORDER actually work ........ + +2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) + | 12 lines Fix greeting retrieval from IMAP Properly check for + the current voicemail state and if it doesn't exist, create it. + (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch + uploaded by mmichelson (license 60) Tested by: jpeeler ........ + +2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 213327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 | + mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 + lines Fix a crash by checking the proper pointer for validity + before deferencing it. (closes issue #15751) Reported by: atis + Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license + 242) ........ + +2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com> + + * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) | + 5 lines Fix compile when certain G711 menuselect options are + enabled. (closes issue #15697) Reported by: slavon ........ + +2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com> + + * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 + (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) + | 8 lines Fixes memory leak caused by incorrectly freeing + mixmonitor (closes issue #15699) Reported by: edantie Patches: + mixmonitor.patch uploaded by edantie (license 862) ........ + ................ + +2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 213098 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | + tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines + Better parsing for the "register" line Allows characters that are + otherwise used as delimiters to be used within certain fields + (like the secret). (closes issue #15008, closes issue #15672) + Reported by: tilghman Patches: 20090818__issue15008.diff.txt + uploaded by tilghman (license 14) Tested by: lmadsen, tilghman + ........ + + * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | + tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines + If we have realtime caching enabled, 'sip reload' must purge + users/peers, even if the config files haven't changed. (closes + issue #12869) Reported by: bcnit Patches: + 20090819__issue12869__2.diff.txt uploaded by tilghman (license + 14) Tested by: lasko ........ + +2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 213046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 | + russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines + Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........ + +2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com> + + * /: Merged revisions 212939 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 | + kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line + Remove some accidentally-committed properties. ........ + + * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /, + UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball: + Merged revisions 212922 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 | + kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6 + lines Convert this branch to Opsound music-on-hold. For more + details: + http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ + ........ + +2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com> + + * /, configs/extconfig.conf.sample: Merged revisions 212857 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18 + Aug 2009) | 4 lines Make the default extconfig.conf match entries + with the sample res_mysql.conf. This eliminates a future source + of possible confusion with the configuration of 1.6.1 and higher. + ........ + +2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 + (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 + Aug 2009) | 1 line Removed some deadwood and added some doxygen + comments. ........ ................ + +2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com> + + * main/manager.c, /: Merged revisions 212764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug + 2009) | 18 lines Merged revisions 212763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug + 2009) | 11 lines Delay the creation of temporary files until we + have a valid manager command to handle. Without this patch, + asterisk creates a temporary file before determining if the + specified command is valid. If invalid, we weren't properly + cleaning up the file. (closes issue #15730) Reported by: zmehmood + Patches: M15730.diff uploaded by junky (license 177) Tested by: + zmehmood ........ ................ + +2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) + | 4 lines Check the return value of opendir(3), or we may crash. + (closes issue #15720) Reported by: tobias_e ........ + +2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com> + + * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug + 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in + chan_agent. (closes issue #15668) Reported by: davidw ........ + + * main/logger.c: Merged revisions 212574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 | + seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8 + lines Correct the return value check for ast_safe_system. The + logic here was reversed as ast_safe_system returns -1 on error + and not on success. Fix suggested by reporter. (closes issue + #15667) Reported by: loic ........ + +2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com> + + * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 + (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) + | 12 lines Fix segfault when reloading chan_misdn. If more ports + were specified than configured in misdn.conf a reload would crash + asterisk. The problem was the unconfigured port was using data + from the previously configured port. When the data for an + unconfigured port was freed a crash would result from the double + free. (closes issue #12113) Reported by: agupta Patches: + bug12113.patch uploaded by jpeeler (license 325) ........ + ................ + +2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 + (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix + uninitialized variable causing random MWI indications. (closes + issue #15727) Reported by: doda Patches: dahdi_changes.patch + uploaded by doda (license 853) ........ r212430 | rmudgett | + 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix + uninitialized variable. ........ ................ + +2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 | + tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines + Add SSL_VERIFYPEER, as requested on the -users list ........ + +2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 | + kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3 + lines Ensure that T38FaxVersion is put into outgoing SDP in the + proper case. ........ + +2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 | + file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines + Check an actual populated variable when seeing if we need to do + video or not. ........ + +2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com> + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13 + Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by: + suretec ........ + +2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug + 2009) | 17 lines Merged revisions 211953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug + 2009) | 10 lines This patch adds additional checking when + generating queue log TRANSFER events. The additional checks + prevent generation of false TRANSFER events in certain + situations. (closes issue #14536) Reported by: aragon Patches: + queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) + Tested by: aragon, mnicholson ........ ................ + + * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 | + mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 + lines Make asterisk handle 423 Interval Too Short messages + better. This change uses separate values for the acceptable + minimum expiry provided by the 423 error and the expiry value + stored in the configuration file. Previously, the value pulled + from the configuration file would be overwritten. (closes issue + #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff + uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch + uploaded by Nick (license 657) Tested by: mnicholson ........ + +2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com> + + * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12 + Aug 2009) | 33 lines Added three new attributes and applied a + patch to res_config_ldap.c attributetype ( + AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC + 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR + caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) + attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC + 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR + caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) + attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' + DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch + SUBSTR caseIgnoreSubstringsMatch SYNTAX + 1.3.6.1.4.1.1466.115.121.1.15) and patch + fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) + Reported by: macogeek Patches: + fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license + 863) Tested by: suretec ........ + +2009-08-10 19:51 +0000 [r211580-211585] 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 + ........ ................ + + * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c, + res/res_config_curl.c, channels/chan_usbradio.c, + channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c, + apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c, + main/asterisk.c, main/dsp.c, main/timing.c, + doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c, + utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c, + cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c, + apps/app_followme.c, main/enum.c, main/indications.c, + res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c, + main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c, + funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c, + funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, + res/res_config_ldap.c, apps/app_adsiprog.c, + funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c, + funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c, + apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c, + codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c, + funcs/func_cut.c, channels/chan_oss.c, main/netsock.c, + apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c, + pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c, + apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c, + apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /, + apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c, + res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c, + main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, + main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c, + main/features.c, main/http.c, channels/xpmr/xpmr.c, + apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c, + channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c, + apps/app_disa.c, main/acl.c, apps/app_originate.c, + channels/iax2-provision.c: AST-2009-005 + +2009-08-10 14:15 +0000 [r211350] 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 [r211235-211278] 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 20:18 +0000 [r211122] 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 18:20 +0000 [r211051] 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 13:10 +0000 [r210995] 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 21:47 +0000 [r210911-210917] 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 17:49 +0000 [r210820] 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 20:43 +0000 [r210686] 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 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com> + + * /: 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 ........ ................ + + * 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 14:55 +0000 [r210191-210241] 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 ........ ................ + + * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c, + channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt, + contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c, + channels/chan_skinny.c, configs/mgcp.conf.sample, + doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt, + configs/res_ldap.conf.sample: Merged revisions 210190 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 + Aug 2009) | 11 lines Rename 'canreinvite' option to + 'directmedia', with backwards compatibility. It is clear from + multiple mailing list, forum, wiki and other sorts of posts that + users don't really understand the effects that the 'canreinvite' + config option actually has, and that in some cases they think + that setting it to 'no' will actually cause various other + features (T.38, MOH, etc.) to not work properly, when in fact + this is not the case. This patch changes the proper name of the + option to what it should have been from the beginning + ('directmedia'), but preserves backwards compatibility for + existing configurations. ........ + +2009-08-01 11:33 +0000 [r209837-209906] 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 ........ + ................ + + * /, main/event.c: Merged revisions 209835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 | + russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines + Fix ast_event_queue_and_cache() to actually do the cache() part. + (closes issue #15624) Reported by: ffossard Tested by: russell + ........ + +2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com> + + * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c, + main/pbx.c, /, main/Makefile, channels/misdn/ie.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 22:01 +0000 [r209715] 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 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com> + + * channels/chan_console.c, include/asterisk/abstract_jb.h, + apps/app_forkcdr.c, channels/chan_dahdi.c, + contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, + codecs/lpc10/pitsyn.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 ........ + +2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | + mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 + lines Fix a crash that can result if text codecs are allowed but + textsupport is disabled. (closes issue #15596) Reported by: + fabled Patches: sip-red.patch uploaded by fabled (license 448) + ........ + +2009-07-28 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-beta4 + +2009-07-28 00:19 +0000 [r209328] 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 [r209265-209282] 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 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Merged revisions 209235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 | + mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 + lines Gracefully handle malformed RTP text packets. AST-2009-004 + ........ + +2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com> + + * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, + channels/chan_vpb.cc, res/res_smdi.c, /, + include/asterisk/module.h, main/features.c, res/res_agi.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 20:23 +0000 [r209135-209222] 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 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com> + + * apps/app_rpt.c, 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 15:40 +0000 [r209059] 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 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_iax2.c, /, main/translate.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 ........ ................ + +2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info> + + * contrib/scripts/install_prereq, /: Merged revisions 208886 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 + Jul 2009) | 2 lines add OpenBSD to the install_prereq script + ........ + +2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info> + + * contrib/scripts/install_prereq, /: Merged revisions 208848 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 + Jul 2009) | 2 lines libxml2-dev is needed as well by default. + ........ + + * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions + 208813 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 | + mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 + lines add default alias reload to run module reload. Requiring + 'module reload' to reload everything, including core etc makes + russell very unhappy. The default configuration already loads the + 'friendly' aliases template. Added 'reload=module reload' to that + template. Also removed the comment in main/cli.c that reload + should come back. ........ + +2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_iax2.c, /, channels/chan_skinny.c, + main/translate.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 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 | + russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines + Remove trailing whitespace. ........ + + * main/cli.c, /: Merged revisions 208706 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 | + russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines + Note that "reload" needs to be added back. I keep getting annoyed + at having to type "module reload" to reload everything, so I'm + adding a note that we need to add "reload" back. "module reload" + doesn't really make sense as the command to reload everything, + including the core. ........ + + * main/cli.c, /: Merged revisions 208693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 | + russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines + Don't log a warning for something that does not affect operation. + ........ + +2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com> + + * /: Fixing trunk-blocked property. + +2009-07-24 18:56 +0000 [r208596] 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 18:32 +0000 [r208591] 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 15:06 +0000 [r208551] 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. ........ + +2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info> + + * contrib/scripts/install_prereq, /: Merged revisions 208542 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 + Jul 2009) | 13 lines use aptitude for debian based systems The + function to check wether we need to install packages was using + dpkg-query which was gives wrong output on Debian 5 Also, the + apt-get has been replaced with aptitude because aptitude is now + the preferred way to handle packages on Debian (closes issue + #15570) Reported by: mvanbaak Patches: + 2009072400_installprereq-aptitude.diff uploaded by mvanbaak + (license 7) ........ + +2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/frame.h, 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: 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 19:36 +0000 [r208391] 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 19:25 +0000 [r208387] 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 [r208266-208320] 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 ........ ................ + +2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com> + + * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 | + qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines + Restore an int declaration on PPC platforms. This x is one crafty + little bugger... It was used for 2 different things (one of which + was only done on PPC) in 1.4. One of the uses were removed in + trunk, and with it went the declaration. (closes issue #14038) + Reported by: ffloimair ........ + +2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 | + tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines + Clarify documentation on 'realtime update2' to show more than one + condition. (closes issue #15357) Reported by: snuffy Patches: + bug_fix_doc_update2.diff uploaded by snuffy (license 35) + (slightly modified by me) ........ + + * /, 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 22:23 +0000 [r207930] Russell Bryant <russell@digium.com> + + * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009) + | 4 lines Note that we use tabs instead of spaces for + indentation. I'm surprised this was never actually in here... + ........ + +2009-07-21 20:30 +0000 [r207785-207862] 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 r207638, this approach could + potentially block for an unacceptable amount of time. + +2009-07-21 14:32 +0000 [r207727] 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 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com> + + * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile, + codecs/Makefile, utils/Makefile, funcs/Makefile, + codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile, + codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, + pbx/Makefile, res/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 04:51 +0000 [r207638] 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 22:14 +0000 [r207523] 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 16:41 +0000 [r207364] 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 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009) + | 3 lines Flag field in wrong position. Reported by "Hoggins!" on + asterisk-dev list. ........ + +2009-07-18 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: 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 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com> + + * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17 + Jul 2009) | 2 lines Add flag here, too (as requested by jsmith) + ........ + + * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 + Jul 2009) | 2 lines Document the "flag" field in the + voicemessages table. ........ + +2009-07-17 19:40 +0000 [r207104-207159] 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 17:52 +0000 [r206869-207030] 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 ........ ................ + + * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) + | 6 lines TIMEOUT(absolute) returned negative value. (closes + issue #15513) Reported by: ys ........ + + * 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 16:53 +0000 [r206811] 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 22:04 +0000 [r206770] 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 21:50 +0000 [r206765] 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 20:20 +0000 [r206703] 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 16:04 +0000 [r206639] 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 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged + 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 19:49 +0000 [r206565] 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 14:59 +0000 [r206389] 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 01:59 +0000 [r206373] 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 + ........ ................ + +2009-07-13 23:27 +0000 [r206281] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 | + dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines + dns lookup of peername rather than peer's host in + transmit_register() (closes issue #15052) Reported by: fsantulli + Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by + fsantulli (license 818) Tested by: fsantulli ........ + +2009-07-13 16:24 +0000 [r206187] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009) + | 2 lines Remove reference to non-existent help file ........ + +2009-07-10 21:46 +0000 [r205986] 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 18:45 +0000 [r205942] 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 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com> + + * /, 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 16:47 +0000 [r205841] 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 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 | + kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11 + lines Eliminate extraneous LOG_DEBUG messages generated by + app_fax. The transmit_audio() and transmit_t38() functions in + app_fax have processing loops that are supposed to wait for + frames to arrive on the channel and then handle them, but they + also have short timeouts so that the loops can have watchdog + timers and do other required processing. This commit changes the + loops to not actually call ast_read() and attempt to process the + returned frame unless a frame actually arrived, eliminating + hundreds of LOG_DEBUG messages and slightly improving + performance. ........ + +2009-07-10 15:58 +0000 [r205779] 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 15:36 +0000 [r205773] 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 23:56 +0000 [r205731] 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 21:27 +0000 [r205699] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c: + 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 16:19 +0000 [r205595-205603] 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. ........ ................ + + * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /: + 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 08:34 +0000 [r205535] 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 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com> + + * include/asterisk/pbx.h, include/asterisk/devicestate.h, + main/pbx.c, /, main/devicestate.c: 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. ........ + ................ + + * /, main/devicestate.c: Merged revisions 205410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 | + dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines + missing comma in devstatestring array ........ + +2009-07-08 19:28 +0000 [r205353] 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 18:22 +0000 [r205302] 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 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com> + + * /, main/features.c: Merged revisions 205254 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 | + dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines + Fixes Park() argument handling Park() was not respecting the + arguments passed to it. Any extension/context/priority given to + it was being ignored. This patch remedies this. (closes issue + #15380) Reported by: DLNoah ........ + +2009-07-08 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com> + + * main/say.c: oops, fixing build + +2009-07-08 16:55 +0000 [r205217] 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 16:30 +0000 [r205207] 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 15:57 +0000 [r205148-205154] 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. ........ + + * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c, + /, main/Makefile, res/res_crypto.c, main/ssl.c (added): 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. ........ + +2009-07-06 13:41 +0000 [r204951] 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 22:06 +0000 [r204838] 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 16:12 +0000 [r204711] 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/ ........ ................ + +2009-06-30 21:30 +0000 [r204611] 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 18:55 +0000 [r204478] 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 [r204473] 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 ........ + ................ + +2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com> + + * configs/res_config_sqlite.conf (removed), + configs/res_config_sqlite.conf.sample (added), /: Merged + revisions 204440 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 | + russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines + Rename res_config_sqlite.conf to res_config_sqlite.conf.sample + (missing .sample). ........ + +2009-06-29 22:53 +0000 [r204250-204304] 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 09:55 +0000 [r203961] Russell Bryant <russell@digium.com> + + * CHANGES, /: Merged revisions 203960 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 | + russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines + Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. + ........ + +2009-06-27 01:24 +0000 [r203941] 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 22:14 +0000 [r203857] 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 21:27 +0000 [r203782-203828] 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 20:18 +0000 [r203731] 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 19:49 +0000 [r203715] Russell Bryant <russell@digium.com> + + * include/asterisk/devicestate.h, main/pbx.c, /, + main/devicestate.c: Merged revisions 203702 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | + russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines + Make invalid hints report Unavailable instead of Idle. (closes + issue #14413) Reported by: pj ........ + +2009-06-26 19:48 +0000 [r203712] 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 19:42 +0000 [r203709] 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 ........ + +2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com> + + * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, + main/channel.c, main/frame.c, /, channels/chan_sip.c, + apps/app_fax.c: 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. ........ + +2009-06-25 21:46 +0000 [r203445] 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 21:21 +0000 [r203400] 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 21:08 +0000 [r203379] 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 ........ ................ + +2009-06-25 19:27 +0000 [r203276] Jason Parker <jparker@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | + 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) + event. This would occasionally cause one-way audio when using + hardware DTMF detection. (closes issue #14761) Reported by: + tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) + Tested by: tzafrir, dimas ........ + +2009-06-25 16:08 +0000 [r203119] Russell Bryant <russell@digium.com> + + * /, 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 21:27 +0000 [r203077] 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 18:30 +0000 [r202970] 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 18:11 +0000 [r202928] 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 23:58 +0000 [r202842] Sean Bright <sean@malleable.com> + + * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions + 202840-202841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 | + seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1 + line Remove some trailing whitespace before making content + changes. ........ r202841 | seanbright | 2009-06-23 19:57:07 + -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in + the CDR tex documentation. ........ + +2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com> + + * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009) + | 2 lines Clean up section hierarchy for the CDR chapter. + ........ + +2009-06-23 22:12 +0000 [r202765] 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 16:33 +0000 [r202673] 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 20:19 +0000 [r202495-202511] 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 ........ ................ + + * /, 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 16:31 +0000 [r202473] 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 15:37 +0000 [r202411] David Vossel <dvossel@digium.com> + + * main/loader.c, /, include/asterisk/module.h: Merged revisions + 202410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | + dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines + attempting to load running modules Modules placed in the priority + heap for loading were not properly removed from the linked list. + This resulted in some modules attempting to load twice. ........ + +2009-06-22 15:17 +0000 [r202340-202346] 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 16:16 +0000 [r202261-202265] 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 [r202186] 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 21:08 +0000 [r202007] 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 20:26 +0000 [r201995] 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 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009) + | 4 lines Fix 2 typos and add support for wide character types. + Reported by Benny Amorsen via the asterisk-users mailing list. + http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html + ........ + + * /, 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 16:44 +0000 [r201679] David Vossel <dvossel@digium.com> + + * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c, + utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c, + utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c, + pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c, + main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /, + channels/h323/ast_h323.cxx: 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 15:40 +0000 [r201614] 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) + ........ ................ + +2009-06-18 15:23 +0000 [r201595] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 | + dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines + parsing extension correctly from sip register lines If a + transport type was specified, but no extension, parsing of the + extension would return whatever was after the transport rather + than defaulting to 's'. (closes issue #15111) Reported by: ffs + Patches: chan_sip.c_register-parser.patch uploaded by ffs + (license 730) Tested by: ffs, dvossel ........ + +2009-06-17 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) + | 7 lines Initialize additional variables, to prevent a possible + crash. (closes issue #15186) Reported by: ajohnson Patches: + 20090528__issue15186.diff.txt uploaded by tilghman (license 14) + Tested by: ajohnson ........ + +2009-06-17 20:12 +0000 [r201461-201465] 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 20:01 +0000 [r201447-201454] David Vossel <dvossel@digium.com> + + * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | + dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines + ast_channel_datastore_alloc is no longer used. updating + datastores.txt to reflect that. ........ + + * 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 19:49 +0000 [r201446] 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 15:25 +0000 [r201360] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | + dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines + SIP registry ref count error During a sip reload, the list of + sip_registry objects are supposed to be traversed, unlinked, and + destroyed, but destruction never takes place due to a ref + counting error. This causes a memory leak when registry items are + removed from sip.conf and reloaded. While the registries are + removed from the global list, they are not removed from the + scheduler. Because of this, SIP register attempts continue to be + sent out for the item even though it may no longer be in the + .conf. (closes issue #15295) Reported by: amorsen Review: + https://reviewboard.asterisk.org/r/282/ ........ + +2009-06-17 12:06 +0000 [r201265] 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 22:30 +0000 [r201224] 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 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/frame.h, apps/app_chanspy.c, + apps/app_mixmonitor.c, main/channel.c, main/autoservice.c, + main/frame.c, /, apps/app_meetme.c, main/slinfactory.c, + include/asterisk/linkedlists.h, main/file.c, + include/asterisk/channel.h: Merged revisions 201056 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/ ........ + ................ + + * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged + revisions 201090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ 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. ........ + + * /, 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 16:28 +0000 [r200984] 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 16:05 +0000 [r200948] 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 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com> + + * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 | + eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines + Show the interface name on error, if it is not found. If the + smdiport specified is not found, show the interface name instead + of '(null)'. ........ + +2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 200799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 | + moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep + backwards compatible chan_dahdi with older openr2 versions by not + using the new skip category feature unless supported ........ + +2009-06-16 01:30 +0000 [r200690-200765] 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: Merged revisions 200689 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 | + kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11 + 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 15:23 +0000 [r200517] 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 ........ ................ + +2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, + build_tools/menuselect-deps.in: Merged revisions 200477 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun + 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit + in menuselect made me realize this was never done but was working + anyways also added support for skip category request feature of + openr2 and updated chan_dahdi.conf.sample ........ + +2009-06-12 19:08 +0000 [r200364] 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 ........ ................ + +2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com> + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu, + 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting + run directory group ownership. (issue #13153) Reported by: + pabelanger ........ + + * 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#. ........ + + * /: Undo block of revision 199782 (will be merging it momentarily) + +2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com> + + * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null + +2009-06-11 21:18 +0000 [r200154] 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 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-beta3 + +2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com> + + * build_tools/make_version_h, /, build_tools/make_version_c: 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 20:37 +0000 [r199998] 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 20:18 +0000 [r199966] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | + mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 + lines Only try to use the invite_branch on outgoing INVITEs with + auth credentials. I have added a comment to the code to help ease + understanding of the logic here as well. ........ + +2009-06-10 16:13 +0000 [r199860] Sean Bright <sean.bright@gmail.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 20:48 +0000 [r199744-199819] 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 ........ + + * main/loader.c, /, res/res_timing_pthread.c, + include/asterisk/module.h, res/res_timing_dahdi.c, + res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) + | 11 lines module load priority This patch adds the option to + give a module a load priority. The value represents the order in + which a module's load() function is initialized. The lower the + value, the higher the priority. The value is only checked if the + AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER + flag is not set, the value will never be read and the module will + be given the lowest possible priority on load. Since some modules + are reliant on a timing interface, the timing modules have been + given a high load priorty. (closes issue #15191) Reported by: + alecdavis Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/262/ ........ + +2009-06-08 19:39 +0000 [r199634] Sean Bright <sean.bright@gmail.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. ........ ................ + +2009-06-08 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Recorded merge of revisions 199588 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, + 08 Jun 2009) | 9 lines Fix a deadlock that could occur when + setting rtp stats on SIP calls. (closes issue #15143) Reported + by: cristiandimache Patches: 15143.patch uploaded by mmichelson + (license 60) Tested by: cristiandimache ........ + +2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 199368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 | + russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines + Switch from "echo -n" to printf. On my mac, the -n was just + getting printed out. ........ + +2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com> + + * include/asterisk/devicestate.h, /, main/devicestate.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 13:52 +0000 [r199230] 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 19:15 +0000 [r199140] 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 ........ + ................ + +2009-06-04 14:53 +0000 [r199054] Sean Bright <sean.bright@gmail.com> + + * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /: + Merged revisions 199051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun + 2009) | 47 lines Merged revisions 199022 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun + 2009) | 40 lines Safely handle AMI connections/reload requests + that occur during startup. During asterisk startup, a lock on the + list of modules is obtained by the primary thread while each + module is initialized. Issue 13778 pointed out a problem with + this approach, however. Because the AMI is loaded before other + modules, it is possible for a module reload to be issued by a + connected client (via Action: Command), causing a deadlock. The + resolution for 13778 was to move initialization of the manager to + happen after the other modules had already been lodaded. While + this fixed this particular issue, it caused a problem for users + (like FreePBX) who call AMI scripts via an #exec in a + configuration file (See issue 15189). The solution I have come up + with is to defer any reload requests that come in until after the + server is fully booted. When a call comes in to ast_module_reload + (from wherever) before we are fully booted, the request is added + to a queue of pending requests. Once we are done booting up, we + then execute these deferred requests in turn. Note that I have + tried to make this a bit more intelligent in that it will not + queue up more than 1 request for the same module to be reloaded, + and if a general reload request comes in ('module reload') the + queue is flushed and we only issue a single deferred reload for + the entire system. As for how this will impact existing + installations - Before 13778, a reload issued before module + initialization was completed would result in a deadlock. After + 13778, you simply couldn't connect to the manager during startup + (which causes problems with #exec-that-calls-AMI configuration + files). I believe this is a good general purpose solution that + won't negatively impact existing installations. (closes issue + #15189) (closes issue #13778) Reported by: p_lindheimer Patches: + 06032009_15189_deferred_reloads.diff uploaded by seanbright + (license 71) Tested by: p_lindheimer, seanbright Review: + https://reviewboard.asterisk.org/r/272/ ........ ................ + +2009-06-03 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com> + + * main/channel.c, /, main/features.c, include/asterisk/channel.h: + Merged revisions 198856 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | + dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines + Generic call forward api, ast_call_forward() The function + ast_call_forward() forwards a call to an extension specified in + an ast_channel's call_forward string. After an ast_channel is + called, if the channel's call_forward string is set this function + can be used to forward the call to a new channel and terminate + the original one. I have included this api call in both + channel.c's ast_request_and_dial() and feature.c's + feature_request_and_dial(). App_dial and app_queue already + contain call forward logic specific for their application and + options. (closes issue #13630) Reported by: festr Review: + https://reviewboard.asterisk.org/r/271/ ........ + + * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) + | 8 lines fixes issue with channels not going down after transfer + Iax2 currently does not support native bridging if the timeoutms + value is set. We check for that in iax2_bridge, but then set + timeoutms to 0 by default. If the timeoutms is not provided it is + set to -1. By setting timeoutms to 0 it is processed causing a + bridging retry loop. (closes issue #15216) Reported by: oxymoron + Tested by: dvossel ........ + +2009-06-02 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 198791 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | + file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines + Correct documentation for the register line, specifically where + the domain should be specified. (closes issue #14367) Reported + by: Nick_Lewis ........ + +2009-06-01 21:04 +0000 [r198730] Russell Bryant <russell@digium.com> + + * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) + | 2 lines Tell the IAX2 parser about more control frame types. + ........ + +2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/scripts/meetme.sql: Merged revisions 198626 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 + Jun 2009) | 2 lines Add information for new meetme realtime + fields ........ + +2009-05-31 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) + | 2 lines Fix documentation for FIELDQTY. ........ + +2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com> + + * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | + 11 lines Avoid a crash when res_timing_dahdi is unloaded but + wasn't properly loaded. if dahdi_test_timer() fails, + timing_funcs_handle remains NULL causing a crash when calling + ast_unregister_timing_interface() with a NULL pointer. (closes + issue #15234) Reported by: eliel Patches: timing_dahdi1.diff + uploaded by eliel (license 64) ........ + +2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com> + + * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) + | 12 lines Merged revisions 198311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) + | 5 lines Fix a crash that occurred when MWI SMDI messages + expired. (closes issue #14561) Reported by: cmoss28 ........ + ................ + +2009-05-30 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com> + + * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | + seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 + lines Properly terminate the receive buffer before sending to + iksemel. aji_io_recv takes the maximum number of bytes to read + (instead of the total buffer size), so we have to subtract 1 from + our buffer size. Without this, when we receive packets that are + larger than our buffer, iksemel will choke and things get wonky. + (closes issue #15232) Reported by: lp0 Patches: + 05302009_res_jabber.c.patch uploaded by seanbright (license 71) + Tested by: seanbright, lp0 ........ + + * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May + 2009) | 19 lines Merged revisions 198370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May + 2009) | 12 lines Properly terminate AMI JabberSend response + messages. The response message (either Error or Success) needs an + extra trailing \r\n after the fields to inform the client that + the message is complete. (closes issue #14876) Reported by: srt + Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright + (license 71) asterisk_14876.patch uploaded by srt (license 378) + trunk-14876-2.diff uploaded by phsultan (license 73) ........ + ................ + + * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May + 2009) | 15 lines Merged revisions 198251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May + 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we + treat a missing one. (closes issue #15056) Reported by: + p_lindheimer Patches: 05292009_bug15056.diff uploaded by + seanbright (license 71) Tested by: p_lindheimer ........ + ................ + +2009-05-30 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | + file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines + When removing all packets from a dialog we also need to free the + data if present. ........ + +2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant <russell@digium.com> + + * /, configs/modules.conf.sample: Merged revisions 198186 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 + May 2009) | 2 lines Suggesting that only a single timing module + be loaded is no longer necessary. ........ + + * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) + | 2 lines Improve handling of trying to ACK too many timer + expirations. ........ + + * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) + | 38 lines Resolve issues with choppy sound when using + res_timing_pthread. The situation that caused this problem was + when continuous mode was being turned on and off while a rate was + set for a timing interface. A very easy way to replicate this bug + was to do a Playback() from behind a Local channel. In this + scenario, a rate gets set on the channel for doing file playback. + At the same time, continuous mode gets turned on and off about + every 20 ms as frames get queued on to the PBX side channel from + the other side of the Local channel. Essentially, this module + treated continuous mode and a set rate as mutually exclusive + states for the timer to be in. When I dug deep enough, I observed + the following pattern: 1) Set timer to tick every 20 ms. 2) Wait + almost 20 ms ... 3) Continuous mode gets turned on for a queued + up frame 4) Continuous mode gets turned off 5) The timer goes + back to its tick per 20 ms. state but starts counting at 0 ms. 6) + Goto step 2. Sometimes, res_timing_pthread would make it 20 ms + and produce a timer tick, but not most of the time. This is what + produced the choppy sound (or sometimes no sound at all). Now, + the module treats continuous mode and a set rate as completely + independent timer modes. They can be enabled and disabled + independently of each other and things work as expected. (closes + issue #14412) Reported by: dome Patches: issue14412.diff.txt + uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt + uploaded by russell (license 2) Tested by: DennisD, russell + ........ + +2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com> + + * CREDITS, /: Merged revisions 198083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 | + eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines + Apply anti-spam obfuscation to an email address. ........ + +2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com> + + * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged + revisions 198072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May + 2009) | 21 lines Merged revisions 198068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May + 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as + the default CDR disposition. This change also involves the + addition of an AST_CDR_FLAG_ORIGINATED flag that is used on + originated channels to distinguish: them from dialed channels. + (closes issue #12946) Reported by: meral Patches: null-cdr2.diff + uploaded by mnicholson (license 96) Tested by: mnicholson, + dbrooks (closes issue #15122) Reported by: sum Tested by: sum + ........ ................ + +2009-05-29 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com> + + * /, main/file.c: Merged revisions 198064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 | + file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix + a memory leak of the write buffer when writing a file. ........ + +2009-05-29 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com> + + * Makefile, /: Merged revisions 198000 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May + 2009) | 15 lines Merged revisions 197998 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May + 2009) | 8 lines Fix 'make config' target for Slackware. There was + a missing semi-colon after the echo statement in the Makefile + that was causing problems for some users. Fix suggested by + reporter. (closes issue #15225) Reported by: pdavis ........ + ................ + +2009-05-29 16:29 +0000 [r197994] Russell Bryant <russell@digium.com> + + * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) + | 2 lines Trim trailing whitespace so that I can work on this bug + without it bothering me. :-) ........ + +2009-05-28 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com> + + * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) + | 8 lines Update documentation in MixMonitor. Updated the + MixMonitor documentation for the 'b' option so that it is more + obvious that you must not optimize away the Local channel when + using this option. (closes issue #14829) Reported by: licedey + Tested by: mmichelson, licedey, lmadsen ........ + +2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 + lines Fix a bug where the trunkmtu setting was not set to the + default value of 1240 on load but was on reload. ........ + +2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com> + + * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | + 19 lines Merged revisions 197562 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | + 13 lines Use the address we already know when reloading a peer + with nat=yes. If we already have an address for a peer, and we + are reloading the sip configuration, try to use that address to + contact the peer, instead of getting it from the Contact. (closes + issue #15194) Reported by: ibc Patches: sip.patch uploaded by + eliel (license 64) Tested by: manwe ........ ................ + +2009-05-28 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: + Merged revisions 197606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May + 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, + 28 May 2009) | 16 lines Allow for media to arrive from an + alternate source when responding to a reinvite with 491. When we + receive a SIP reinvite, it is possible that we may not be able to + process the reinvite immediately since we have also sent a + reinvite out ourselves. The problem is that whoever sent us the + reinvite may have also sent a reinvite out to another party, and + that reinvite may have succeeded. As a result, even though we are + not going to accept the reinvite we just received, it is + important for us to not have problems if we suddenly start + receiving RTP from a new source. The fix for this is to grab the + media source information from the SDP of the reinvite that we + receive. This information is passed to the RTP layer so that it + will know about the alternate source for media. Review: + https://reviewboard.asterisk.org/r/252 ........ ................ + + * main/audiohook.c, apps/app_chanspy.c, /, + include/asterisk/audiohook.h: Merged revisions 197543 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 + (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May + 2009) | 21 lines Add flags to chanspy audiohook so that audio + stays in sync. There are two flags being added to the chanspy + audiohook here. One is the pre-existing + AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that + the read and write slinfactories on the audiohook do not skew + beyond a certain tolerance. In addition, there is a new audiohook + flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, + we do not allow for a slinfactory to build up a substantial + amount of audio before flushing it. For this particular issue, + this means that the person spying on the call will hear the + conversations in real time with very little delay in the audio. + (closes issue #13745) Reported by: geoffs Patches: 13745.patch + uploaded by mmichelson (license 60) Tested by: snblitz ........ + ................ + +2009-05-28 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com> + + * /, main/utils.c: Merged revisions 197538 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | + file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix + a bug in stringfields where it did not actually free the pools of + memory. (closes issue #15074) Reported by: pj ........ + + * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | + 15 lines Merged revisions 197466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 + lines Fix a bug where the flag indicating the presence of rport + would get overwritten by the nat setting. The presence of rport + is now stored as a separate flag. Once the dialog is setup and + authenticated (or it passes through unauthenticated) the proper + nat flag is set. (closes issue #13823) Reported by: dimas + ........ ................ + +2009-05-28 11:40 +0000 [r197441] Gavin Henry <ghenry@suretecsystems.com> + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif, doc/ldap.txt, + configs/res_ldap.conf.sample: issue #15155 and issue #15156 from + trunk + +2009-05-27 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com> + + * /, main/xml.c: Merged revisions 197374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 | + tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines + Revert commit 192032. This define is needed on Mac OS X. ........ + +2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May + 2009) | 3 lines Ensure that this header includes xmldoc.h, since + it depends on it. ........ + +2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com> + + * Makefile, /: Merged revisions 197260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 | + seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 + lines Use bash explicitly when calling build_tools/mkpkgconfig + from the Makefile. Since we use bashisms in + build_tools/mkpkgconfig, we should call on bash explicitly when + running from the Makefile, otherwise we get errors during a 'make + install.' (closes issue #15209) Reported by: seandarcy ........ + +2009-05-27 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_cut.c: Recorded merge of revisions 197209 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500 + (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) + | 5 lines Use a different determinator on whether to print the + delimiter, since leading fields may be blank. (closes issue + #15208) Reported by: ramonpeek Patch by me, though inspired in + part by a patch from ramonpeek ........ ................ + +2009-05-27 17:28 +0000 [r197176] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, include/asterisk/channel.h: Fix broken attended + transfers The bridge was terminating immediately after the + attended transfer was completed. The problem was because upon + reentering ast_channel_bridge nexteventts was checked to see if + it was set and if so could possibly return AST_BRIDGE_COMPLETE. + (closes issue #15183) Reported by: andrebarbosa Tested by: + andrebarbosa, tootai, loloski + +2009-05-27 16:12 +0000 [r196950-197092] Sean Bright <sean.bright@gmail.com> + + * configs/smdi.conf.sample, configs/extensions.conf.sample, + configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /, + configs/vpb.conf.sample: Merged revisions 197089 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May + 2009) | 6 lines Fix references to /etc/dahdi/system.conf and + /etc/asterisk/chan_dahdi.conf in the sample configuration files. + (closes issue #15207) Reported by: seandarcy ........ + + * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May + 2009) | 9 lines Display an error message when chan_alsa fails to + load due to a missing or inaccessible configuration file. Before + this change, when chan_alsa failed to load due to a missing or + inaccessible configuration file, no message would be displayed. + With this change, when chan_alsa fails to load due to a missing + or inaccessible configuration file, a message will be displayed. + (closes issue #14760) Reported by: Nick_Lewis Patches: + chan_alsa.c-confload.patch uploaded by Nick (license 657) + ........ + + * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 | + seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8 + lines Reset the terminal to the correct fg/bg after XML + documenation is rendered. (closes issue #15200) Reported by: + ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright + (license 71) Tested by: ajohnson ........ + + * main/manager.c, /: Merged revisions 196945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 | + seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13 + lines Add ActionID to CoreShowChannel event. There is + inconsistency in how we handle manager responses that are lists + of items and, unfortunately, third parties have come to rely on + ActionID being on every event within those lists instead of just + keeping track of the ActionID for the current response. This + change makes CoreShowChannels include the ActionID with each + CoreShowChannel event generated as a result of it being called. + (closes issue #15001) Reported by: sum Patches: + patchactionid2.patch uploaded by sum (license 766) ........ + +2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com> + + * /, autoconf/ast_check_osptk.m4 (added), configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 196946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 | + russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines + Update configure script to check for OSP toolkit 3.5.0. (closes + issue #14988) Reported by: tzafrir Patches: configure.ac.diff + uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded + by homesick (license 91) ........ + + * /, res/res_convert.c: Merged revisions 196843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) + | 16 lines Merged revisions 196826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) + | 9 lines Resolve a file handle leak. The frames here should have + always been freed. However, out of luck, there was never any + memory leaked. However, after file streams became reference + counted, this code would leak the file stream for the file being + read. (closes issue #15181) Reported by: jkroon ........ + ................ + +2009-05-26 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com> + + * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 | + seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2 + lines Add a missing unref for queues in handle_statechange. + ........ + +2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | + file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix + a bug where the sip unregister CLI command did not completely + unregister the peer. (closes issue #15118) Reported by: alecdavis + Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis + (license 585) ........ + + * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, + 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 + lines Remove some bash specific stuff from safe_asterisk. (closes + issue #10812) Reported by: paravoid Patches: + safe_asterisk_bashism.diff uploaded by tzafrir (license 46) + ........ ................ + +2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1 + line set MFCR2_CATEGORY just when starting the pbx ........ + +2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 196416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | + dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines + SIP set outbound transport type from Registration In sip.conf the + transport option allows for the configuration of what transport + types (udp, tcp, and tls) a peer will accept, but only the first + type listed was used for outbound connections. This patch changes + this. Now the default transport type is only used until the peer + registers. When registration takes place the transport type is + parsed out of the Contact header. If the Contact header's + transport type is equal to one that the peer supports, the peer's + default transport type for outbound connections is set to match + the Contact header's type. If the Contact header's transport type + is not present, then the peer's default transport type is set to + match the one the peer registered with. When a peer unregisters + or the registration expires, the default transport type for that + peer is reset. (closes issue #12282) Reported by: rjain Patches: + reg_patch_1.diff uploaded by dvossel (license 671) Tested by: + dvossel (closes issue #14727) Reported by: pj Patches: + reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, + dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ + +2009-05-22 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com> + + * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 | + eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines + Unregister every registered application by MiniVM. The MinivmMWI + application was not being unregistered on unload and we were not + able to load again the module or reload it. (closes issue #15174) + Reported by: junky Patches: unregister_minivm_mwi.diff uploaded + by junky (license 177) ........ + +2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, + 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 + lines Fix a bug where using immediate with mISDN caused a cause + code of 16 to get sent back instead of 1 if the 's' extension did + not exist. (closes issue #12286) Reported by: lmamane ........ + ................ + +2009-05-21 19:15 +0000 [r196000] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 + (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) + | 14 lines Sign problem calculating timestamp for iax frame leads + to no audio on the receiving peer. There are rare cases in which + a frame's delivery timestamp is slightly less than the iax2_pvt's + offset. This causes the pvt's timestamp to be a small negative + number, but since the timestamp value is unsigned it looks like a + huge positive number. This patch checks for this negative case + and sets the ms to zero. A similar check is already done right + below this one in the 'else' statement. (closes issue #15032) + Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp + uploaded by guillecabeza (license 380) Tested by: guillecabeza + (closes issue #14216) Reported by: Andrey Sofronov ........ + ................ + +2009-05-21 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com> + + * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 + (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May + 2009) | 13 lines This commit prevents cdr records with + AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated + in certain cases. This is accomplished by adding two functions to + update the answer time and disposition of calls that checks for + the proper lock flags. These functions are used in the + ast_bridge_call() function so that ForkCDR(A) calls are + respected. This patch also modifies the way ast_bridge_call() + chooses the cdr record to base the bridged_cdr on. Previously the + first unlocked cdr record would be chosen, now instead the first + cdr record is chosen and forked cdr records are moved to the + bridge_cdr. This allows the original cdr record and any forked + cdr records to be properly updated with answer and end times. + (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes + issue #14744) Reported by: deepesh ........ ................ + +2009-05-20 23:31 +0000 [r195842] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 | + tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines + If a variable had a blank value upon the initial setting, then it + would do nothing. Identified by Dmitry Andrianov via private + email, fixed by me. ........ + +2009-05-20 17:35 +0000 [r195639-195707] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 195698 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | + 12 lines Merged revisions 195688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 + lines Fix some code that wrongly assumed a pointer would always + be non-NULL when dealing with CDRs after a bridge. (closes issue + #15079) Reported by: barryf ........ ................ + + * /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | + 12 lines Merged revisions 195635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 + lines Fix a bug where the MeetMe option 'D' did not actually + prompt for the pin. (closes issue #15050) Reported by: pmhaddad + ........ ................ + +2009-05-19 20:19 +0000 [r195531] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500 + (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) + | 7 lines Ensure thread keys are initialized before attempting to + access them. (closes issue #14889) Reported by: jaroth Patches: + app_voicemail.c.patch uploaded by msirota (license 758) Tested + by: msirota, BlargMaN ........ ................ + +2009-05-19 14:49 +0000 [r195452] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | + 14 lines Merged revisions 195448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 + lines Fix a bug where direct RTP setup would partially occur even + when disabled if the calling channel was answered. (issue #13545) + Reported by: davidw (issue #14244) Reported by: mbnwa ........ + ................ + +2009-05-18 21:25 +0000 [r195405] Eliel C. Sardanons <eliels@gmail.com> + + * main/manager.c, /: Merged revisions 195369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 | + eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines + Fix the CLI command 'manager show command' documentation and + functionality. The CLI command 'manager show command' supports + passing multiple action names in the same line, but it was not + allowing that because of a incorrect check in the argumentes + counter. Also the documentation was updated to show that this + usage of the command is possible. ........ + +2009-05-18 20:55 +0000 [r195359-195373] Tilghman Lesher <tlesher@digium.com> + + * apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c, + apps/app_voicemail.c, res/res_smdi.c, /, + include/asterisk/monitor.h: Merged revisions 195370 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 + (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) + | 8 lines Add a similar dependency on SMDI for voicemail as + already exists for ADSI. (closes issue #14846) Reported by: pj + Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman + (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by + tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt + uploaded by tilghman (license 14) ........ ................ + + * main/asterisk.c, /: Merged revisions 195320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 | + tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines + Move the spawn of astcanary down, until after the call to + daemon(3). This avoids possible conflicts with the internal + implementation of daemon(3). (closes issue #15093) Reported by: + tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by + tilghman (license 14) Tested by: tzafrir ........ + +2009-05-18 19:01 +0000 [r195319] Mark Michelson <mmichelson@digium.com> + + * apps/app_externalivr.c, /: Merged revisions 195316 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May + 2009) | 18 lines Fix externalivr's setvariable command so that it + properly sets multiple variables. The command had a for loop that + was guaranteed to only execute once since the continuation + operation of the loop would set the input buffer NULL. I rewrote + the loop so that its operation was more obvious, and it would set + multiple variables correctly. I also reduced stack space required + for the function, constified the input string, and modified the + function so that it would not modify the input string while I was + at it. (closes issue #15114) Reported by: chris-mac Patches: + 15114.patch uploaded by mmichelson (license 60) Tested by: + chris-mac ........ + +2009-05-18 15:57 +0000 [r195212] Joshua Colp <jcolp@digium.com> + + * main/frame.c, /: Merged revisions 195207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) | + 14 lines Merged revisions 195206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 + lines Fix a typo which caused loss of audio when using G729 in + some scenarios with a smoother present. (closes issue #15105) + Reported by: bamby Patches: process-vad-correctly.diff uploaded + by bamby (license 430) ........ ................ + +2009-05-18 14:54 +0000 [r195164] Eliel C. Sardanons <eliels@gmail.com> + + * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged + revisions 195162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 | + eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines + Warn about the use of the application WaitExten() within a + Macro(). Update applications documentation to warn the user about + the use of the WaitExten() application within a Macro(). + Recommend the use of Read() instead. (closes issue #14444) + Reported by: ewieling ........ + +2009-05-18 14:00 +0000 [r195099] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 195096 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | + 12 lines Merged revisions 195095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 + lines Fix a bug where the codecs of the called party leg were not + properly sent back to the caller call leg when reinvited. (closes + issue #13569) Reported by: bkw918 ........ ................ + +2009-05-18 13:50 +0000 [r195093-195094] Eliel C. Sardanons <eliels@gmail.com> + + * /, main/xml.c: Merged revisions 195075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 | + eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do + not avoid loading the XML documentation if not XInclude + substitution is done. ........ + + * doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions + 194982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 | + eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines + Allow to include sections of other parts of the xml + documentation. Avoid duplicating xml documentation by allowing to + include other parts of the xml documentation using XInclude. + Example: <xi:include + xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" /> + (Insert this line to include the synopsis of the CHANNEL function + xml documentation). It is also possible to include documentation + from other files in the 'documentation/' directory using the + href="" attribute inside a xinclude element. (closes issue + #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling + ........ + +2009-05-18 13:39 +0000 [r195092] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | + file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix + a bug where specifying an empty outboundproxy would cause packets + to get sent to ourself. (closes issue #15106) Reported by: + timeshell ........ + +2009-05-18 13:14 +0000 [r195024] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 195021 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009) + | 12 lines Recorded merge of revisions 195020 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) + | 5 lines Don't try to unlock a bogus channel. (closes issue + #15144) Reported by: cristiandimache ........ ................ + +2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons <eliels@gmail.com> + + * main/pbx.c, /: Merged revisions 194945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 | + eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines + Fix a missing unlock in case of error, and a missing free(). + Always free the allocated memory for a string field, because we + are always using it (not only when xmldocs are enabled). Also if + there is an error allocating memory for the string field remember + to unlock the list of registered applications, before returning. + ........ + +2009-05-15 22:48 +0000 [r194836-194877] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 + (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) + | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to + terminate invalid registrations. Instead it sent another REGAUTH + if the authentication challenge failed. This caused a loop of + REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) + (closes issue #14867) Reported by: aragon Tested by: dvossel + (closes issue #14717) Reported by: mobeck Patches: + regauth_loop_update_patch.diff uploaded by dvossel (license 671) + Tested by: dvossel ........ ................ + + * channels/chan_iax2.c, channels/iax2-parser.c, + channels/iax2-parser.h, /, channels/iax2.h: Merged revisions + 194833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) + | 24 lines Merged revisions 194557,194685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) + | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue + where people are reporting "Ghost" channels in their 'iax2 show + channels' output. The confusion is caused by channels being + listed as "(NONE)" with format "unknown". These are not channels + of coarse. They are usually just pending registration or poke + requests, but it is confusing output. To help make sense of this + I have added two columns to 'iax2 show channels'. One shows the + first message which started the transaction, and the second shows + the last message sent by either side of the call. This helps + diagnose why the entry exists and why it may not go away. (closes + issue #14207) Reported by: clive18 Review: + https://reviewboard.asterisk.org/r/246/ ........ r194685 | + dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines + Update to previous IAX2 "Ghost" Channels patch. Fixed some + comments made on reviewboard for the previous patch. (issue + #14207) ........ ................ + +2009-05-15 18:44 +0000 [r194717-194768] Russell Bryant <russell@digium.com> + + * configs/logger.conf.sample, /: Merged revisions 194765 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r194765 | russell | 2009-05-15 13:43:42 -0500 + (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) + | 2 lines Fix some spelling fail. ........ ................ + + * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged + revisions 194722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 | + russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines + Shuttle some bits around to address some gain issues with G.722. + (closes AST-209) ........ + + * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged + revisions 194718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 | + russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines + Further simplify codec_g722 build. ........ + + * codecs/Makefile, /: Merged revisions 194714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 | + russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines + Actually force running make for g722. ........ + +2009-05-15 13:47 +0000 [r194650] Michiel van Baak <michiel@vanbaak.info> + + * CREDITS, /: Merged revisions 194649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 | + mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines + add eliel ........ + +2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons <eliels@gmail.com> + + * doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May + 2009) | 16 lines Allow to specify an enumlist inside an enum. It + was not possible to use an enumlist inside an enum: <enumlist> + <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist> + Now we will be able to insert as many levels as we want. (closes + issue #15112) Reported by: lmadsen ........ + +2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming <kpfleming@digium.com> + + * /: Merged revisions 194520 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May + 2009) | 9 lines Merged revisions 194509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May + 2009) | 1 line Update URL to Reviewboard ........ + ................ + +2009-05-14 22:23 +0000 [r194510] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May + 2009) | 30 lines Merged revisions 194484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May + 2009) | 24 lines Fix a race condition where a reinvite could + trigger a 482 response. The loop detection/spiral detection code + in chan_sip used the owner channel's state as a criterion for + determining if the incoming INVITE is a looped request. The + problem with this is that the INVITE-handling code happens in a + different thread than the thread that marks the owner channel as + being up. As a result, if a reinvite were to come in very + quickly, say from another Asterisk on the same LAN, it was + possible for the reinvite to arrive before the owner channel had + been set to the up state. This patch corrects the problem by + using the invitestate of the sip_pvt instead, since that can be + guaranteed to be set correctly by the time the reinvite arrives. + Since there is a switch statement further in the INVITE-handling + code, the AST_STATE_RINGING state also checks the invitestate of + the sip_pvt in case we should actually be treating the channel as + if it were up already. (closes issue #12215) Reported by: jpyle + Patches: 12215_confirmed.patch uploaded by mmichelson (license + 60) Tested by: lmadsen ........ ................ + +2009-05-14 17:07 +0000 [r194437] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 | + file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix + a bug where the 'T' option to Meetme did not work. (closes issue + #15031) Reported by: Stochastic (closes issue #13801) Reported + by: justdave ........ + +2009-05-14 16:23 +0000 [r194431] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 194430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 | + tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines + If the timing ended on a zero, then we would loop forever. + (closes issue #14983) Reported by: teox Patches: + 20090513__issue14983.diff.txt uploaded by tilghman (license 14) + Tested by: teox ........ + +2009-05-13 13:42 +0000 [r194213] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 194209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | + 18 lines Merged revisions 194208 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | + 11 lines Fix RFC2833 issues with DTMF getting duplicated and with + duration wrapping over. (closes issue #14815) Reported by: + geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) + Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue + #14460) Reported by: moliveras Tested by: moliveras ........ + ................ + +2009-05-13 00:54 +0000 [r194141] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 194138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009) + | 14 lines Merged revisions 194137 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) + | 7 lines Fix logic for how to proceed with a single digit + extension. (closes issue #15091) Reported by: andrew Patches: + 20090512__issue15091.diff.txt uploaded by tilghman (license 14) + Tested by: andrew ........ ................ + +2009-05-12 22:48 +0000 [r194059] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May + 2009) | 22 lines Merged revisions 194028 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May + 2009) | 16 lines This change modifies app_queue to properly + generate CDR records in failure situations. This involves setting + a proper cdr disposition coresponding to the given failure + condition and ensuring the proper information is stored in the + cdr record. (closes issue #13691) Reported by: dferrer Tested by: + mnicholson (closes issue #13637) Reported by: atis Tested by: + atis ........ ................ + +2009-05-12 20:51 +0000 [r193962] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | + mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 + lines Update spiral support in trunk and 1.6.X to match what is + in 1.4. In 1.4, a SIP spiral is treated the same way as a call + forward. This works much better than what is currently in trunk + and 1.6.X. The code in trunk and 1.6.X did not create a new call + to the recipient of the spiral, instead trying to continue the + same call. In addition to just being plain wrong, this also had + the side effect of only being able to spiral calls to other SIP + channels. With this in place, as long as call forwards are + honored, SIP spirals will work properly. This means that it will + work for outbound calls made by the Queue, Dial, and Page + applications. For originated calls and spool calls, however, the + spiral will not work properly until a generic call forward + mechanism is introduced into Asterisk. (relates to issue #13630) + ........ + +2009-05-12 20:42 +0000 [r193823-193959] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500 + (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) + | 6 lines Avoid initializing routines if the authentication + fails. Fixes a crash (RR) issue. (closes issue #14508) Reported + by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by + tiziano (license 377) ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009) + | 2 lines Convert a THREADSTORAGE object into a simple malloc'd + object (as suggested by Russell on -dev) ........ + + * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500 + (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) + | 18 lines Move 300 bytes around on the stack, to make more room + for an extension buffer. This allows more concurrent extensions + to be copied for a single voicemail, without creating a + possibility of upsetting existing users, where a dialplan could + run out of stack space where it had run fine before. + Alternatively, we could have allocated off the heap, but that is + a larger change and would have increased the chance for + instability introduced by this change. This is really solved + starting in 1.6.0.11, as the use of an ast_str buffer allows an + unlimited number of extensions (up to available memory). We + additionally create a new warning message when the buffer length + is exceeded, permitting administrators to see an issue after the + fact, whereas previously the list was silently truncated. (closes + issue #14739) Reported by: p_lindheimer Patches: + 20090417__bug14739.diff.txt uploaded by tilghman (license 14) + Tested by: p_lindheimer ........ ................ + +2009-05-11 22:12 +0000 [r193719] Russell Bryant <russell@digium.com> + + * /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009) + | 12 lines Fix some timer state corruption. In res_timer_timerfd, + handle the case that set_rate gets called while a timer is still + in continuous mode. In this case, we want to remember the + configured rate, but not actually set it until continuous mode + has been disabled. Thanks to dvossel for finding and helping to + debug the problem. (closes issue #15080) Reported by: dvossel + Tested by: dvossel ........ + +2009-05-11 19:17 +0000 [r193617] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 + (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) + | 12 lines Sent wrong message to clear a call we started if the + other end has not responed yet. In the state MISDN_CALLING (i.e. + SETUP was sent but no answer has arrived yet), it is not allowed + to clear the call with RELEASE_COMPLETE. It must be cleared with + DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a + SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: + chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 + ........ ................ + +2009-05-11 18:59 +0000 [r193612] Leif Madsen <lmadsen@digium.com> + + * /, funcs/func_channel.c: Update CHANNEL(transfercapabilities) + documentation. (closes issue #15073) Reported by: pkempgen + Patches: 20090511__issue15073__trunk.diff.txt uploaded by + tilghman (license 14) + +2009-05-10 17:08 +0000 [r193503] Joshua Colp <jcolp@digium.com> + + * main/bridging.c, /: Merged revisions 193502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 | + file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix + a bug where receiving a control frame of subclass -1 would cause + certain channels to get hung up. ........ + +2009-05-09 11:33 +0000 [r193462] Russell Bryant <russell@digium.com> + + * include/asterisk/event.h, /: Merged revisions 193461 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009) + | 2 lines Minor documentation update for ast_event_queue(). + ........ + +2009-05-08 20:52 +0000 [r193390] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | + dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines + TCP not matching valid peer. find_peer() does not find a valid + peer when using pvt->recv as the sockaddr_in argument. Because of + the way TCP works, the port number in pvt->recv is not what we're + looking for at all. There is currently only one place that + find_peer searches for a peer using the sockaddr_in argument. If + the peer is not found after using pvt->recv (works for UDP since + the port number will be correct), a temp sockaddr_in struct is + made using the Contact header in the sip_request. This has the + correct port number in it. Review: + http://reviewboard.digium.com/r/236/ ........ + +2009-05-08 19:51 +0000 [r193350] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 193349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 | + mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12 + lines Reset the members' call counts when resetting queue + statistics. This helps to prevent odd scenarios where a queue + will claim to have taken 0 calls, but the members appear to have + taken a non-zero amount. (closes issue #15068) Reported by: sum + Patches: patchreset.patch uploaded by sum (license 766) Tested + by: sum ........ + +2009-05-08 15:36 +0000 [r193336] Sean Bright <sean.bright@gmail.com> + + * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May + 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate + CLI completion. ........ + +2009-05-08 14:55 +0000 [r193266] David Vossel <dvossel@digium.com> + + * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 + (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) + | 9 lines "misdn show config" segfaults asterisk, if no MSN lists + (closes issue #14976) Reported by: alecdavis Patches: + misdn_config.diff.txt uploaded by alecdavis (license 585) Tested + by: alecdavis, FabienToune ........ ................ + +2009-05-08 14:12 +0000 [r193197] Kevin P. Fleming <kpfleming@digium.com> + + * configs/logger.conf.sample, /, main/logger.c: Merged revisions + 193194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May + 2009) | 13 lines Merged revisions 193193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May + 2009) | 7 lines Make absolute paths for logger channels work + properly (Note: This is not a new feature, it was previously + undocumented and broken.) The Asterisk logger has a feature to + support absolute pathnames for logger channels, but the code + implementing the feature was broken. This has been fixed, and the + absolute path feature is now documented in the sample + logger.conf. ........ ................ + +2009-05-07 23:44 +0000 [r193123] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 193120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) + | 26 lines Merged revisions 193119 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) + | 19 lines Fix Background within a Macro for FreePBX. If the + single digit DTMF is an extension in the specified context, then + go there and signal no DTMF. Otherwise, we should exit with that + DTMF. If we're in Macro, we'll exit and seek that DTMF as the + beginning of an extension in the Macro's calling context. If + we're not in Macro, then we'll simply seek that extension in the + calling context. Previously, someone complained about the + behavior as it related to the interior of a Gosub routine, and + the fix (#14011) inadvertently broke FreePBX (#14940). This + change should fix both of these situations, but with the possible + incompatibility that if a single digit extension does not exist + (but a longer extension COULD have matched), it would have + previously gone immediately to the "i" extension, but will now + need to wait for a timeout. (closes issue #14940) Reported by: + p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by + tilghman (license 14) Tested by: p_lindheimer ........ + ................ + +2009-05-07 22:51 +0000 [r193080] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 + (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) + | 5 lines Give a more helpful message when an incoming call's + dialed extension does not match. Added the dialed extension and + context to the chan_misdn messages warning that the dialed number + cannot be matched in the dialplan. ........ ................ + +2009-05-07 17:53 +0000 [r192936-193008] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 | + tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines + Second result should not contain data from the first result. + (closes issue #15039) Reported by: jims Patches: + 20090506__issue15039.diff.txt uploaded by tilghman (license 14) + Tested by: jims ........ + + * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) + | 6 lines Send DTMF frame before playing back audio. (closes + issue #14858) Reported by: barryf Patches: + 20090507__bug14858.diff.txt uploaded by tilghman (license 14) + ........ + + * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) + | 17 lines Merged revisions 192932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) + | 10 lines Eliminate repetition of fullcontact during + reconstruction. If the fullcontact field appears in both the + sippeers and the sipregs table, then during reconstruction of the + field, it will otherwise be doubled. (closes issue #14754) + Reported by: Alexei Gradinari Patches: + 20090506__bug14754.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen ........ ................ + +2009-05-07 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-beta2 + +2009-05-06 22:20 +0000 [r192874] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 192861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) + | 17 lines Merged revisions 192858 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) + | 10 lines Make ParkedCall application stop execution of the + dialplan after hang up Just changed park_exec to always return + non-zero. I really wasn't entirely sure at first if this was a + bug. Decided it was since it would be surprising when not using + ParkedCall in the dialplan to hang up and have dialplan execution + continue. (closes issue #14555) Reported by: francesco_r ........ + ................ + +2009-05-06 17:57 +0000 [r192813] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | + 1 line Make sure that we do not clear the down flag on the BRI + during PTMP link transients. Also refix SS7 audio that the early + media patch broke. ........ + +2009-05-06 17:41 +0000 [r192637-192810] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | + 10 lines Fix a bug where a timer would be created but not + acknowledged. This scenario crept up if chan_iax2 was loaded with + no configuration file present. It would create a timer and tell + it to go at an interval but the thread that normally acknowledges + it would not be created because no configuration file was + present. The timer will now be closed if no configuration file is + present. (closes issue #15014) Reported by: madkins ........ + + * res/res_clialiases.c, /: Merged revisions 192736 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4 + lines Make the code that prevents an infinite loop from happening + into a case insensitive check. (thanks eliel) ........ + + * res/res_clialiases.c, /: Merged revisions 192700 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5 + lines Fix an infinite loop with tab completion of CLI aliases + that reference themselves. (closes issue #15020) Reported by: + junky ........ + + * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | + 14 lines Merged revisions 192633 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 + lines Update some old logic to stop both begin and end DTMF + frames from reaching the core if rfc2833 is not enabled. (closes + issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded + by dimas (license 88) ........ ................ + +2009-05-05 20:02 +0000 [r192528] Sean Bright <sean.bright@gmail.com> + + * /, static-http/astman.js: Merged revisions 192525 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400 + (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May + 2009) | 11 lines Fix Javascript error when using astman.js in + Internet Explorer. Internet Explorer (tested with 7.0) does not + like trailing commas on constructs like object initializers, so + get rid of them to avoid some errors. (closes issue #15026) + Reported by: rajnishgiri Patches: bug15026.patch uploaded by + seanbright (license 71) Tested by: seanbright ........ + ................ + +2009-05-05 18:27 +0000 [r192402-192480] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 192462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | + 15 lines Merged revisions 192454 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 + lines Fix an incorrect assumption that certain values on the + channel will always exist when they may not. The CDR code + involved with bridges wrongly assumed that the currently + executing application and data values will always exist. It is + possible for this to be false when call forwarding is involved. + (closes issue #14984) Reported by: gincantalupo ........ + ................ + + * apps/app_followme.c, /: Merged revisions 192430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) | + 12 lines Merged revisions 192429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 + lines Fix a bug where the followme application would continue + trying numbers after the caller hung up. (closes issue #13624) + Reported by: sgenyuk ........ ................ + + * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | + file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines + Fix a bug with setting t38pt_udptl at the user or peer level. If + an incoming call authenticated as a user or peer and t38pt_udptl + was not set to yes in general then no UDPTL session would be + present and any T38 related things would fail. This commit + changes it so that if after authenticating T38 is enabled but no + UDPTL session is present one will be created. (issue AST-215) + ........ + +2009-05-05 13:43 +0000 [r192298-192360] Kevin P. Fleming <kpfleming@digium.com> + + * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c: + Merged revisions 192357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | + kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 + lines Correct some flaws in the memory accounting code for + stringfields and ao2 objects Under some conditions, the memory + allocation for stringfields and ao2 objects would not have + supplied valid file/function names for MALLOC_DEBUG tracking, so + this commit corrects that. ........ + + * main/astobj2.c, main/datastore.c, main/channel.c, /, + include/asterisk/astobj2.h, include/asterisk/datastore.h, + include/asterisk/channel.h: Merged revisions 192318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May + 2009) | 5 lines Properly account for memory allocated for + channels and datastores As in previous commits, when channels are + allocated (with ast_channel_alloc) or datastores are allocated + (with ast_datastore_alloc) properly account for the memory being + owned by the caller, instead of the allocator function itself. + ........ + + * include/asterisk/stringfields.h, /, main/utils.c: Merged + revisions 192279 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | + kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 + lines Ensure that string pools allocated to hold stringfields are + properly accounted in MALLOC_DEBUG mode This commit modifies the + stringfield pool allocator to remember the 'owner' of the + stringfield manager the pool is being allocated for, and ensures + that pools allocated in the future when fields are populated are + owned by that file/function. ........ + +2009-05-04 22:48 +0000 [r192217] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 + (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) + | 11 lines global mohinterpret setting is ignored mohinterpret + and mohsuggest global variables were not copied over during + build_users and build_peers. (closes issue #14728) Reported by: + dimas Patches: v1-14728.patch uploaded by dimas (license 88) + Tested by: dimas, dvossel ........ ................ + +2009-05-04 19:34 +0000 [r192175] Kevin P. Fleming <kpfleming@digium.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 192059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 | + kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5 + lines Ensure that astobj2 memory allocations are properly + accounted for when MALLOC_DEBUG is used This commit ensures that + all astobj2 allocated objects are properly accounted for in + MALLOC_DEBUG mode by passing down the file/function/line + information from the module/function that actually called the + astobj2 allocation function. ........ + +2009-05-04 19:31 +0000 [r192135-192173] Tilghman Lesher <tlesher@digium.com> + + * /, configure, res/res_agi.c: Merged revisions 192171 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009) + | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher. + (closes issue #14985) Reported by: nikkk Patches: + 20090428__bug14985.diff.txt uploaded by tilghman (license 14) + 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license + 14) Tested by: nikkk ........ + + * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 + May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes + issue #14671) Reported by: Chainsaw Patches: + asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by + Chainsaw (license 723) ........ + +2009-05-04 17:45 +0000 [r192097] Leif Madsen <lmadsen@digium.com> + + * apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 | + lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines + Commit documentation changes related to issue #14801. (issue + #14801) ........ + +2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons <eliels@gmail.com> + + * /, main/xml.c: Merged revisions 192032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 | + eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do + not re-define _POSIX_C_SOURCE if it was already defined. ........ + +2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming <kpfleming@digium.com> + + * /, configs/modules.conf.sample: Merged revisions 191955 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 + May 2009) | 8 lines Ensure that by default only one console + channel driver is loaded This configuration file was changed to + ensure that only one console channel driver (chan_oss) is loaded + by default, but the change would only work if chan_console was + not built. Now it will work as expected; if chan_alsa or + chan_console are built and installed, they will not be loaded + unless explicity requested. ........ + +2009-05-03 14:06 +0000 [r191885] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 191884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 | + russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines + Remove unnecessary compiler flag ........ + +2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/logger.c: Merged revisions 191775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 | + kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5 + lines Fix an error in queue_log file rotation optimization code + This code was copy-and-pasted without properly changing + references to event_rotate into queue_rotate, so under some + conditions the log rotation would rotate queue_log even though it + was not necessary. ........ + +2009-05-02 15:52 +0000 [r191703] Sean Bright <sean.bright@gmail.com> + + * main/asterisk.c, /: Merged revisions 191700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 | + seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 + line Update copyright year to 2009 ........ + +2009-05-01 20:02 +0000 [r191554-191563] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) + | 13 lines Merged revisions 191559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) + | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. + (closes issue #14993) Reported by: BigJimmy Patches: causepatch + uploaded by BigJimmy (license 371) ........ ................ + + * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) + | 4 lines Set debug message back to DEBUG level. (closes issue + #15007) Reported by: hulber ........ + +2009-05-01 18:20 +0000 [r191508] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 191489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) + | 15 lines Merged revisions 191488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) + | 9 lines Fix DTMF not being sent to other side after a partial + feature match This fixes a regression from commit 176701. The + issue was that ast_generic_bridge never exited after the feature + digit timeout had elapsed, which prevented the queued DTMF from + being sent to the other side. This issue was reported to me + directly. ........ ................ + +2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Merged revisions 191367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | + tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines + Detect eaccess (or euidaccess) before using it. Reported by + Andrew Lindh via the -dev list. ........ + + * main/asterisk.c, /: Merged revisions 191283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 | + tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 + lines Change working directory to / under certain conditions. If + backgrounding and no core will be produced, then changing the + directory won't break anything; likewise, if the CWD isn't + accessible by the current user, then a core wasn't possible + anyway. (closes issue #14831) Reported by: chris-mac Patches: + 20090428__bug14831.diff.txt uploaded by tilghman (license 14) + 20090430__bug14831.diff.txt uploaded by tilghman (license 14) + Tested by: chris-mac ........ + + * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged + revisions 191219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | + tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines + Make H.323 compile with FDLEAK detection code enabled ........ + +2009-04-29 18:40 +0000 [r191139] David Brooks <dbrooks@digium.com> + + * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 | + dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines + Removing crufty code that is no longer necessary. Code cleanup. + ........ + +2009-04-29 08:59 +0000 [r190994] Russell Bryant <russell@digium.com> + + * main/indications.c, /: Merged revisions 190993 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 | + russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines + Log an error message if indications.conf is not found. (closes + issue #14990) Reported by: tzafrir Patches: indications_err.diff + uploaded by tzafrir (license 46) ........ + +2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development <support@transnexus.com> + + * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr + 2009) | 2 lines Updated for OSP Toolkit 3.5. ........ + +2009-04-28 17:33 +0000 [r190907] Tilghman Lesher <tlesher@digium.com> + + * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009) + | 2 lines UniqueID column has a maximum size of 150 ........ + +2009-04-28 14:17 +0000 [r190732-190869] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 190865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 | + kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5 + lines Build XML documention from *only* the source files that + have docs in them Change the build process so that + doc/core-en_US.xml is dependent solely on the source files that + have documentation in them, not on all source files. ........ + + * /, Makefile.rules: Merged revisions 190861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 | + kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5 + lines Remove Makefile rules for bison and flex sources We never, + ever want these files to processed automatically, because we + store the output files in Subversion and users should never need + to rebuild them. ........ + + * /, configure, include/asterisk/autoconfig.h.in: Merged revisions + 190725 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr + 2009) | 13 lines Merged revisions 190721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr + 2009) | 7 lines Fix 'inconsistent line endings' when autoconf + 2.63 is used Attempt to make configure script regeneration 'safe' + using autoconf 2.63, which embeds a bare CR into the script, thus + making Subversion complain about inconsistent line endings This + commit changes the MIME type of the configure script to be + 'binary' thus making Subversion no longer inspect line endings, + and as a bonus 'svn diff' will no longer try to generate diff + output for it, which is not generally useful anyway. ........ + ................ + +2009-04-27 19:36 +0000 [r190729] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 190726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 | + tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines + Don't warn on pipe in the System call. (closes issue #14979) + Reported by: pj ........ + +2009-04-27 19:15 +0000 [r190666] Russell Bryant <russell@digium.com> + + * res/res_smdi.c, /: Merged revisions 190663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) + | 22 lines Merged revisions 190661-190662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) + | 9 lines Resolve a crash in res_smdi when used with chan_dahdi. + When chan_dahdi goes to get an SMDI message, it provides no + search criteria. It just grabs the next message that arrives. + This code was written with the SMDI dialplan functions in mind, + since that is now the preferred method of using SMDI. However, + this broke support of it being used from chan_dahdi. (closes + AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500 + (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........ + ................ + +2009-04-27 16:28 +0000 [r190625] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 190622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 | + mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3 + lines Update warning message to not have pipes and contain all + options. ........ + +2009-04-23 21:23 +0000 [r190383] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ ........ + +2009-04-23 20:44 +0000 [r190355] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 190352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 | + tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines + Labels are sometimes (most of the time?) NULL for extensions. + (closes issue #14895) Reported by: chris-mac Patches: + 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen ........ + +2009-04-23 19:18 +0000 [r190297] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, /: Merged revisions 190287 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, + 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 + lines Fix a bug in chan_local glare hangup detection. If both + sides of a Local channel were hung up at around the same time it + was possible for one thread to destroy the local private + structure and have the other thread immediately try to remove the + already freed structure from the local channel list. ........ + ................ + +2009-04-23 17:47 +0000 [r190253] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 190250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 | + mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9 + lines Fix reversed behavior of leavewhenempty option in + queues.conf. (closes issue #14650) Reported by: alecdavis + Patches: 14650.patch uploaded by mmichelson (license 60) Tested + by: mmichelson, lmadsen ........ + +2009-04-22 21:43 +0000 [r190096] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Merged revisions 190093 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500 + (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) + | 7 lines Detect availability of pthread_rwlock_timedwrlock() + before using it. (closes issue #14930) Reported by: tilghman + Patches: 20090420__bug14930.diff.txt uploaded by tilghman + (license 14) Tested by: mvanbaak, tilghman ........ + ................ + +2009-04-22 21:18 +0000 [r189997-190066] Jeff Peeler <jpeeler@digium.com> + + * main/cli.c, funcs/func_groupcount.c, /, main/app.c, + include/asterisk/channel.h: Merged revisions 190057 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) + | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to + be a bug with old versions of g++ that doesn't allow a structure + member to use the name list. Rename list member to group_list in + ast_group_info and change the few places it is used. (closes + issue #14790) Reported by: stuarth ........ + + * channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx, + channels/chan_h323.c: Merged revisions 189993 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 | + jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines + Make chan_h323 respect packetization settings and fix small + reload issue. Previously, packetization settings were ignored and + now they are not. A new config option 'autoframing' has been + added to mirror the way chan_sip handles it. Turning on the + autoframing option (available both as a global option or per + peer) overrides the local settings with the remote packetization + settings. Testing was performed with varying packetization levels + with the following codecs: ulaw, alaw, gsm, and g729. Also, an + unrelated config reload issue has been fixed in the case of the + config file not changing. (closes issue #12415) Reported by: pj + Patches: 2009012200_h323packetization.diff.txt uploaded by + mvanbaak (license 7), modified by me ........ + +2009-04-22 18:01 +0000 [r189986] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 189951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 | + russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines + Fix call parking callback. Pipes -> Commas. ........ + +2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) + | 7 lines Do not continue to receive DTMF, when the channel is + hungup and about to be destroyed. (closes issue #14858) Reported + by: barryf Patches: 20090421__bug14858.diff.txt uploaded by + tilghman (license 14) Tested by: barryf ........ + + * /, configure, configure.ac: Merged revisions 189813 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009) + | 3 lines Detect liblua on SuSE, and add libm for linking for + Fedora. (Reported via the -dev list, Subject: Compiling Asterisk + with LUA) ........ + +2009-04-21 20:45 +0000 [r189775] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 | + dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines + Fixes segfault when switching UDP to TCP in sip.conf after + reload. If transport in sip.conf is switched from UDP to TCP, + Asterisk segfaults right after issuing a sip reload. The problem + is the socket type is changed to TCP but the fd may still be + present for UDP. Later, when the TCP session should be created or + set using an existing one, it isn't because the old file + descriptor is still present. Now every time transport is changed + during a sip.conf reload, the file descriptor is set to -1, + signifying it must be created or found. (closes issue #14727) + Reported by: pj Tested by: dvossel Review: + http://reviewboard.digium.com/r/229/ ........ + +2009-04-20 22:11 +0000 [r189540] Tilghman Lesher <tlesher@digium.com> + + * main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009) + | 3 lines Use nanosleep instead of poll. This is not just because + mmichelson suggested it, but also because Mac OS X puked on my + poll(). ........ + +2009-04-20 21:41 +0000 [r189536] Terry Wilson <twilson@digium.com> + + * apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r189495 | twilson | 2009-04-20 16:24:34 -0500 + (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 + Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL + ........ ................ r189516 | twilson | 2009-04-20 16:29:29 + -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) + | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is + set ........ ................ + +2009-04-20 21:36 +0000 [r189533] Sean Bright <sean.bright@gmail.com> + + * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400 + (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr + 2009) | 13 lines Properly handle @s within hints in AEL. AEL was + not handling the case of a device hint containing an @ symbol, + which caused parking hints (e.g. hint(park:exten@context)) to + error out the parser. This patch makes AEL treat the @ the same + way it treats colon and ampersand now, meaning the characters are + included in verbatim. (closes issue #14941) Reported by: bpgoldsb + Patches: bug14941.patch uploaded by seanbright (license 71) + Tested by: bpgoldsb ........ ................ + +2009-04-20 17:11 +0000 [r189353] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | + file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines + Fix a bug with non-UDP connections that caused dialogs to not get + freed. This issue crept up because of a reference count issue on + non-UDP based dialogs. The dialog reference count was increased + when transmitting a packet reliably but never decreased. This + caused the dialog structure to hang around despite being unlinked + from the dialogs container. (closes issue #14919) Reported by: + vrban ........ + +2009-04-20 14:07 +0000 [r189281] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 189278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr + 2009) | 18 lines Merged revisions 189277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr + 2009) | 12 lines Move the check for chan->fdno == -1 to after the + zombie/hangup check. Many users were finding that their hung up + channels were staying up and causing 100% CPU usage. (issue + #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch + uploaded by mmichelson (license 60) Tested by: falves11, bamby + ........ ................ + +2009-04-18 01:42 +0000 [r189207-189208] David Vossel <dvossel@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 + (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) + | 12 lines National prefix inserted even when caller ID not + available When the caller ID is restricted, the expected behavior + is for the caller id to be blank. In chan_dahdi, the national + prefix is placed onto the callers number even if its restricted + (empty) causing the caller id to be the national prefix rather + than blank. (closes issue #13207) Reported by: shawkris Patches: + national_prefix.diff uploaded by dvossel (license 671) Review: + http://reviewboard.digium.com/r/220/ ........ ................ + + * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 + (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) + | 12 lines Fixed autologoff in agents.conf not working when agent + logs in via AgentLogin app An agent logs in by calling an + extension that calls the AgentLogin app. In agents.conf + ackcall=always is set, so when they get a call they have the + choice to either acknowledge it or ignore it. autologoff=10 is + set as well, so if the agent ignores the call over 10sec one may + assume that the agent should be logged out (and in this case + hungup on as well), but this was not happening. (closes issue + #14091) Reported by: evandro Patches: autologoff.diff uploaded by + dvossel (license 671) Review: + http://reviewboard.digium.com/r/225/ ........ ................ + +2009-04-17 21:56 +0000 [r189140] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 189137 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) + | 17 lines Merged revisions 188833,189134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) + | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. + Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | + rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines + Modifed/added some debug messages. JIRA ABE-1835 ........ + ................ + +2009-04-17 20:21 +0000 [r189105] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | + mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 + lines Prevent a crash when SIP blonde transferring an unbridged + call. If one attempts to use the attended transfer button on a + SIP phone to transfer an unbridged call (such as a call to an + IVR) but hangs up while the target of the transfer is still + ringing, we need to not crash. The problem was that ast_hangup + was called from outside the channel thread. AST-211 ........ + +2009-04-17 19:47 +0000 [r189081] Sean Bright <sean.bright@gmail.com> + + * main/asterisk.c, /: Merged revisions 189077 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | + seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 + line Fix copy/paste error with 'transmit silence' flag. ........ + +2009-04-17 17:31 +0000 [r189068] Matthew Nicholson <mnicholson@digium.com> + + * main/pbx.c, /: Merged revisions 189010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr + 2009) | 12 lines Merged revisions 189009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr + 2009) | 5 lines Make Busy() application set the CDR disposition + to BUSY. (closes issue #14306) Reported by: cristiandimache + ........ ................ + +2009-04-17 14:50 +0000 [r188941-188950] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | + 22 lines Merged revisions 188946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | + 15 lines Fix a bug where a value used to create the channel name + was bogus. This commit fixes the scenario where an incoming call + is authenticated using a peer entry. Previously the channel name + was created using either the username setting from the sip.conf + entry or the IP address that the call came from. Now the channel + name will be created using the peer name itself. This commit will + not change the way the channel name is generated for users or + friends. (closes issue #14256) Reported by: Nick_Lewis Patches: + chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: + Nick_Lewis, file ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, + 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 + lines Fix a situation where the DAHDI channel private structure + lock was not unlocked when it should have been. (issue AST-210) + ........ ................ + +2009-04-16 22:05 +0000 [r188777-188839] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) + | 14 lines Merged revisions 188835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) + | 7 lines Only update realtime, if global option rtupdate != + false (closes issue #14885) Reported by: deepesh Patches: + 20090413__bug14885.diff.txt uploaded by tilghman (license 14) + Tested by: deepesh ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 + (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) + | 4 lines Umask should not be exported into global namespace. + (closes issue #14912) Reported by: jcapp ........ + ................ + +2009-04-15 20:20 +0000 [r188474-188598] Mark Michelson <mmichelson@digium.com> + + * /, main/file.c: Merged revisions 188585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr + 2009) | 13 lines Merged revisions 188582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr + 2009) | 7 lines Update ast_readvideo_callback to match + ast_readaudio_callback. This fixes potential refcount errors that + may occur on ast_filestreams. AST-208 ........ ................ + + * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | + mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 + lines Fix a couple of queue member reference leaks. ........ + +2009-04-14 17:46 +0000 [r188259-188416] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 188413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | + file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix + an incorrect clock rate when sending T140 text. (closes issue + #14029) Reported by: epicac ........ + + * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | + file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix + a bug with the change I made yesterday to outbound proxy support. + Per discussion with oej on IRC we need the actual IP address, not + the outbound proxy IP address, in the sa field. Upon further + inspection this should make the behaviour of all other uses of + the outbound proxy in the code. ........ + +2009-04-14 05:47 +0000 [r188209-188213] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 188210 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | + tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines + As suggested by Russell, warn users when their dialplan arguments + contain pipes, but not commas. ........ + + * /, utils/smsq.c: Merged revisions 188206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | + tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines + Application delimiter is ',', not '|'. (closes issue #14881) + Reported by: stegro Patches: smsq.patch uploaded by stegro + (license 752) ........ + +2009-04-13 19:33 +0000 [r188105] Mark Michelson <mmichelson@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 188102 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr + 2009) | 5 lines Fix another crash related to cached realtime + music on hold. This was another off-by-one problem caused by + moh_register. ........ + +2009-04-13 16:34 +0000 [r188070] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | + file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines + Fix a bug where using an outbound proxy would cause the local + address to be 127.0.0.1. Copy the outbound proxy IP address into + the SIP dialog structure as the IP address we will be sending to. + This has to be done because the logic that determines what local + IP address to use in the SIP messages is not aware of an outbound + proxy being in place. It only knows what IP address we are + sending to. (closes issue #12006) Reported by: mnicholson + ........ + +2009-04-13 14:20 +0000 [r188039] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | + mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 + lines Set all queue variables on both the caller and member + channels. This allows for the variables to be accessed if a + member macro is run. Thanks to Grigoriy Puzankin for bringing + this up on the -dev list. ........ + +2009-04-10 20:28 +0000 [r187916] Jeff Peeler <jpeeler@digium.com> + + * channels/Makefile, /: Merged revisions 187906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | + jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines + Fix module embedding for chan_h323. Include libchanh323.a in the + modules.link file so that all the symbols can be resolved at link + time. (closes issue #11966) Reported by: dome Patches: + issue_11966.patch uploaded by kpfleming (license 421) Tested by: + jpeeler ........ + +2009-04-10 17:31 +0000 [r187769] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/sip-friends.sql, + contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 + (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 + Apr 2009) | 2 lines Add lastms column to the contributed table + designs ........ ................ + +2009-04-10 16:54 +0000 [r187724] Kevin P. Fleming <kpfleming@digium.com> + + * /, build_tools/embed_modules.xml: Merged revisions 187721 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 + Apr 2009) | 5 lines clean up some patterns for files to remove + add embedding support for bridge and test modules ........ + +2009-04-10 16:05 +0000 [r187679] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | + tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines + Ensure pvt is not NULL before dereferencing it. (closes issue + #14784) Reported by: pj ........ + +2009-04-10 16:01 +0000 [r187677] Russell Bryant <russell@digium.com> + + * tests/test_sched.c, tests/test_heap.c, /: Merged revisions 187675 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 + Apr 2009) | 2 lines Disable test modules by default. ........ + +2009-04-10 03:57 +0000 [r187601] Tilghman Lesher <tlesher@digium.com> + + * main/audiohook.c, main/bridging.c, main/channel.c, main/pbx.c, + main/manager.c, /, include/asterisk/linkedlists.h, + main/features.c, main/http.c, main/app.c, + include/asterisk/lock.h: Merged revisions 187599 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) + | 2 lines Modify headers and macros, according to Russell's + suggestions on the -dev list ........ + +2009-04-09 21:09 +0000 [r187564] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merge revision 187488 from trunk. + +2009-04-09 19:29 +0000 [r187531-187546] David Vossel <dvossel@digium.com> + + * main/audiohook.c, /: Merged revisions 186379 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | + dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines + audio_audiohook_write_list() did not correctly update sample size + after ast_translate. audio_audiohook_write_list() did not take + into account that the sample size may change after translation + depending on if the original frame is is 8khz or 16khz. the + sample size is now updated after translating to reflect this + possibility. This caused the audio on the receiving end to sound + terrible. Thanks to jcolp and mmichelson for helping me work this + out. (issue AST-197) ........ + + * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) + | 16 lines Merged revisions 185845 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) + | 10 lines Fixes issue with dropped calles due to re-Invite glare + and re-Invites never executing after a 491 Acknowledgement for + 491 responses were never being processed because it didn't match + our pending invite's seqno. Since the ACK was never processed, + the 491 frame would continue to be retransmitted until eventually + the call was dropped due to max retries. Now during a pending + invite, if we receive another invite, we send an 491 and hold on + to that glare invite's seqno in the "glareinvite" variable for + that sip_pvt struct. When ACK's are received, we first check to + see if it is in response to our pending invite, if not we check + to see if it is in response to a glare invite. In this case, it + is in response to the glare invite and must be dealt with or the + call is dropped. I've changed the wait time for resending the + re-Invite after receving a 491 response to comply with RFC 3261. + Before this patch the scheduled re-Invite would only change a + flag indicating that the re-Invite should be sent out, now it + actually sends it out as well. (closes issue #12013) Reported by: + alx Review: http://reviewboard.digium.com/r/213/ ........ + ................ + +2009-04-09 19:15 +0000 [r187496] Mark Michelson <mmichelson@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 187421,187424 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, + 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using + cached realtime moh. The moh_register function links an mohclass + and then immediately unrefs the class since the container now has + a reference. The problem with using realtime music on hold is + that the class is allocated, registered, and started in one fell + swoop. The refcounting logic resulted in the count being off by + one. The same problem did not happen when using a static config + because the allocation and registration of an mohclass is a + separate operation from starting moh. This also did not affect + non-cached realtime moh because the classes are not registered at + all. I also have modified res_musiconhold to use the _t_ variants + of the ao2_ functions so that more info can be gleaned when + attempting to trace the refcounts. I found this to be incredibly + helpful for debugging this issue and there's no good reason to + remove it. (closes issue #14661) Reported by: sum ........ + r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr + 2009) | 3 lines Use safe macro practices even though they really + aren't necessary. ........ + +2009-04-09 18:55 +0000 [r187051-187487] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /, include/asterisk/linkedlists.h, + include/asterisk/lock.h: Merged revisions 187483 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 + (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) + | 8 lines Race condition between ast_cli_command() and 'module + unload' could cause a deadlock. Add lock timeouts to avoid this + potential deadlock. (closes issue #14705) Reported by: jamessan + Patches: 20090320__bug14705.diff.txt uploaded by tilghman + (license 14) Tested by: jamessan ........ ................ + + * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | + tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines + Allow '/' in username portion of register; this is a regression. + (closes issue #14668) Reported by: Netview ........ + + * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions + 187363 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) + | 10 lines Merged revisions 187362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) + | 3 lines Permit zero-length text messages in SIP. (Related to an + issue posted to the -users list, subject "AEL2, BASE64_DECODE and + hexadecimal") ........ ................ + + * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, + utils/Makefile, include/asterisk.h, /, main/Makefile, + main/file.c, main/astfd.c (added): Merged revisions 187302 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 + (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) + | 3 lines Add debugging mode for diagnosing file descriptor + leaks. (Related to issue #14625) ........ r187301 | tilghman | + 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, + missed this file in the last commit. ........ ................ + + * /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 | + tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines + If the first column is empty, output a delimiter anyway. (closes + issue #14848) Reported by: john8675309 Patches: + 20090408__bug14848.diff.txt uploaded by tilghman (license 14) + Tested by: john8675309 ........ + +2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson <mmichelson@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 187046 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 + (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr + 2009) | 10 lines Fix a small logical error when loading moh + classes. We were unconditionally incrementing the number of + mohclasses registered. However, we should actually only increment + if the call to moh_register was successful. While this probably + has never caused problems, I noticed it and decided to fix it + anyway. ........ ................ + + * main/channel.c, /: Merged revisions 186985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr + 2009) | 30 lines Merged revisions 186984 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr + 2009) | 24 lines Make a couple of changes with regards to a new + message printed in ast_read(). "ast_read() called with no + recorded file descriptor" is a new message added after a bug was + discovered. Unfortunately, it seems there are a bunch of places + that potentially make such calls to ast_read() and trigger this + error message to be displayed. This commit does two things to + help to make this message appear less. First, the message has + been downgraded to a debug level message if dev mode is not + enabled. The message means a lot more to developers than it does + to end users, and so developers should take an effort to be sure + to call ast_read only when a channel is ready to be read from. + However, since this doesn't actually cause an error in operation + and is not something a user can easily fix, we should not spam + their console with these messages. Second, the message has been + moved to after the check for any pending masquerades. ast_read() + being called with no recorded file descriptor should not + interfere with a masquerade taking place. This could be seen as a + simple way of resolving issue #14723. However, I still want to + try to clear out the existing ways of triggering this message, + since I feel that would be a better resolution for the issue. + ........ ................ + +2009-04-08 12:39 +0000 [r186929] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 | + russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines + Update some comments and resolve potential memory corruption in + chan_sip. While browsing chan_sip the other day, I noticed this + dangerous code in dialog_needdestroy(). This function is an + ao2_callback. It is absolutely _not_ okay to unlock the container + from within this function. It's also not clear why it was useful. + Given that it could cause memory corruption, I have removed it. + There was also a TODO comment left describing a potential + implementation of an improvement to the needdestroy handling. I'm + not convinced that what was described is the best choice here, so + I have briefly described the way that this function is used today + that could be improved. ........ + +2009-04-08 05:08 +0000 [r186901] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | + tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines + Add lastms to the require API call. ........ + +2009-04-08 00:10 +0000 [r186836-186845] Mark Michelson <mmichelson@digium.com> + + * formats/format_wav_gsm.c, /, formats/format_wav.c: Merged + revisions 186842 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr + 2009) | 14 lines Merged revisions 186841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr + 2009) | 8 lines Fix a few typos of the word "frequency." (closes + issue #14842) Reported by: jvandal Patches: frequency-typo.diff + uploaded by jvandal (license 413) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | + mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 + lines Fix bad merge from fix for issue 13867. (closes issue + #14686) Reported by: davidw ........ + + * main/channel.c, /: Merged revisions 186833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr + 2009) | 15 lines Merged revisions 186832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr + 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a + p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, + warning sounds will not be properly played to either party of the + bridge. (closes issue #14845) Reported by: adomjan ........ + ................ + +2009-04-07 22:33 +0000 [r186807] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) + | 10 lines Merged revisions 186775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) + | 3 lines Fix Macro documentation to match current (and intended) + behavior. (See -dev mailing list) ........ ................ + +2009-04-07 20:59 +0000 [r186723] Mark Michelson <mmichelson@digium.com> + + * main/manager.c, /: Merged revisions 186720 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr + 2009) | 12 lines Merged revisions 186719 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr + 2009) | 6 lines Ensure that \r\n is printed after the ActionID in + an OriginateResponse. (closes issue #14847) Reported by: kobaz + ........ ................ + +2009-04-03 20:21 +0000 [r186469] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 + (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr + 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not + properly switch formats when requested Don't offer + AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could + provide a slight performance benefit, the translation core in + Asterisk has some flaws when a channel driver offers multiple raw + formats. this fix is much simpler than fixing the translation + core to solve that issue (although that will be done later). + ........ ................ + +2009-04-03 20:05 +0000 [r186449] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 186444,186447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) + | 14 lines Merged revisions 186415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) + | 7 lines Distinguish in a sent email between simple sends and + forwards. (closes issue #11678) Reported by: jamessan Patches: + 20090330__bug11678.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman, lmadsen ........ ................ r186447 | + tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines + Merged revisions 186445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) + | 2 lines Found a conflict in the last commit, due to multiple + targets ........ ................ + +2009-04-03 15:56 +0000 [r186324] Joshua Colp <jcolp@digium.com> + + * include/asterisk/crypto.h, /: Merged revisions 186321 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, + 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 + lines Fix a problem with the crypto variable definitions not + actually being defined properly. (closes issue #14804) Reported + by: jvandal ........ ................ + +2009-04-03 15:19 +0000 [r186302] Tilghman Lesher <tlesher@digium.com> + + * main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009) + | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820) + Reported by: phsultan ........ + +2009-04-03 14:34 +0000 [r186289] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr + 2009) | 20 lines Fix the ability to retrieve voicemail messages + from IMAP. A recent change made interactive vm_states no longer + get added to the list of vm_states and instead get stored in + thread-local storage. In trunk and all the 1.6.X branches, the + problem is that when we search for messages in a voicemail box, + we would attempt to update the appropriate vm_state struct by + directly searching in the list of vm_states instead of using the + get_vm_state_by_imap_user function. This meant we could not find + the interactive vm_state that we wanted. (closes issue #14685) + Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson + (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ + +2009-04-03 02:11 +0000 [r186233] Russell Bryant <russell@digium.com> + + * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) + | 29 lines Merged revisions 186229 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) + | 21 lines Fix a memory leak in cdr_radius. I came across this + while doing some testing of my ast_channel_ao2 branch. After + running a test overnight that generated over 5 million calls, + Asterisk had taken up about 1 GB of my system memory. So, I + re-ran the test with MALLOC_DEBUG turned on. However, it showed + no leaks in Asterisk during the test, even though Asterisk was + still consuming it somehow. Instead, I turned to valgrind, which + when run with --leak-check=full, told me exactly where the leak + came from, which was from allocations inside the radiusclient-ng + library. This explains why MALLOC_DEBUG did not report it. After + a bit of analysis, I found that we were leaking a little bit of + memory every time a CDR record was passed to cdr_radius. I don't + actually have a radius server set up to receive CDR records. + However, I always have my development systems compile and install + all modules. In addition to making sure there are not build + errors across modules, always loading modules helps find bugs + like this, too, so it is strongly recommend for all developers. + ........ ................ + +2009-04-02 22:00 +0000 [r186178] Mark Michelson <mmichelson@digium.com> + + * configs/features.conf.sample, /: Merged revisions 186175 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 + (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr + 2009) | 5 lines Fix instructions in one-step parking comment to + make more sense. Changed a capital K to a lowercase k. ........ + ................ + +2009-04-02 17:28 +0000 [r186111] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 + (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 + Apr 2009) | 3 lines ensure that the buffer passed to + DAHDI_SET_BUFINFO is fully initialized ........ ................ + +2009-04-02 17:14 +0000 [r186022-186063] Tilghman Lesher <tlesher@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 186060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) + | 16 lines Merged revisions 186059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 + (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 + Apr 2009) | 2 lines Fix for AST-2009-003 ........ + ................ ................ + + * main/strings.c, /: Merged revisions 186021 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 | + tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines + Missed a common case for needing to extend the buffer. (closes + issue #14716) Reported by: sum Patches: + 20090402__bug14716.diff.txt uploaded by tilghman (license 14) + Tested by: sum ........ + +2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 + (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr + 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and + DAHDI_GET_PARAMS ioctls were recently corrected to show that they + do, in fact, read data from userspace as part of their work. due + to this fix, valgrind now reports a number of cases where + chan_dahdi passed an uninitialized (or partially) buffer to these + ioctls, which could lead to unexpected behavior. this patch + corrects chan_dahdi to ensure that buffers passed to these ioctls + are always fully initialized. ........ ................ + +2009-04-01 22:44 +0000 [r185947] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/pbx.h, include/asterisk/strings.h, + main/taskprocessor.c, res/res_odbc.c, + include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c, + main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h, + main/ast_expr2f.c: Merged revisions 185912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 | + tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines + Merge changes from str_substitution that are unrelated to that + branch. Included is a small bugfix to an ast_str helper, but most + of these changes are simply doxygen fixes. ........ + +2009-04-01 13:51 +0000 [r185775] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 185772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) + | 14 lines Merged revisions 185771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) + | 6 lines Fix a case where DTMF could bypass audiohooks. This + change fixes a situation where an audiohook that wants DTMF would + not actually get it. This is in the code path where we end DTMF + digit length emulation while handling a NULL frame. ........ + ................ + +2009-03-31 22:38 +0000 [r185667] Kevin P. Fleming <kpfleming@digium.com> + + * utils, /: Merged revisions 185664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | + kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line + ignore copied (generated) file ........ + +2009-03-31 22:13 +0000 [r185472-185605] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 185604 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 | + mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3 + lines Fix trunk's compilation. ........ + + * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar + 2009) | 12 lines Merged revisions 185599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar + 2009) | 6 lines Fix crash that would occur if an empty member was + specified in queues.conf. (closes issue #14796) Reported by: pida + ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 + (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar + 2009) | 8 lines Fix Russian voicemail intro to say the word + "messages" properly. (closes issue #14736) Reported by: chappell + Patches: voicemail_no_messages.diff uploaded by chappell (license + 8) ........ ................ + +2009-03-31 17:51 +0000 [r185428] David Brooks <dbrooks@digium.com> + + * channels/chan_gtalk.c, /: Merged revisions 185363 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 + (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) + | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains + extra whitespaces To drill into the xmpp to find the capabilities + between channels, chan_gtalk calls iks_child() and iks_next(). + iks_child() and iks_next() are functions in the iksemel xml + parsing library that traverse xml nodes. The bug here is that + both iks_child() and iks_next() will return the next iks_struct + node *regardless* of type. chan_gtalk expects the next node to be + of type IKS_TAG, which in most cases, it is, but in this case (a + call being made from the Empathy IM client), there exists + iks_struct nodes which are not IKS_TAG data (they are extraneous + whitespaces), and chan_gtalk doesn't handle that case, so + capabilities don't match, and a call cannot be made. + iks_first_tag() and iks_next_tag(), on the other hand, will not + return the very next iks_struct, but will check to see if the + next iks_struct is of type IKS_TAG. If it isn't, it will be + skipped, and the next struct of type IKS_TAG it finds will be + returned. This assures that chan_gtalk will find the iks_struct + it is looking for. This fix simply changes all calls to + iks_child() and iks_next() to become calls to iks_first_tag() and + iks_next_tag(), which resolves the capability matching. The + following is a payload listing from Empathy, which, due to the + extraneous whitespace, will not be parsed correctly by iksemel: + <iq from='dbrooksjab@235-22-24-10/Telepathy' + to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> + <session xmlns='http://www.google.com/session' + initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' + id='1837267342'> <description + xmlns='http://www.google.com/session/phone'> <payload-type + clockrate='16000' name='speex' id='96'/> <payload-type + clockrate='8000' name='PCMA' id='8'/> <payload-type + clockrate='8000' name='PCMU' id='0'/> <payload-type + clockrate='90000' name='MPA' id='97'/> <payload-type + clockrate='16000' name='SIREN' id='98'/> <payload-type + clockrate='8000' name='telephone-event' id='99'/> </description> + </session> </iq> Review: http://reviewboard.digium.com/r/181/ + ........ ................ + +2009-03-31 14:59 +0000 [r185264] Russell Bryant <russell@digium.com> + + * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | + russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines + Don't free() an astobj2 object. (closes issue #14672) Reported + by: makoto ........ + +2009-03-31 14:11 +0000 [r185200] Joshua Colp <jcolp@digium.com> + + * main/audiohook.c, /: Merged revisions 185197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | + 15 lines Merged revisions 185196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 + lines Fix crash when moving audiohooks between channels. Handle + the scenario where we are called to move audiohooks between + channels and the source channel does not actually have any on it. + (closes issue #14734) Reported by: corruptor ........ + ................ + +2009-03-30 20:52 +0000 [r185128-185129] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged + revisions 185123 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) + | 9 lines Merged revisions 185121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) + | 1 line Update the channel allocation method documentation. + ........ ................ + + * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 + (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) + | 19 lines Make chan_misdn BRI TE side normally defer channel + selection to the NT side. Channel allocation collisions are not + handled by chan_misdn very well. This patch simply avoids the + problem for BRI only. For PRI, allocation collisions are still + possible but less likely since there are simply more channels + available and each end could use a different allocation strategy. + misdn.conf options available: te_choose_channel - Use to force + the TE side to allocate channels. method - Specify the channel + allocation strategy. (closes issue #13488) Reported by: + Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich + Tested by: crich, siepkes, festr ........ ................ + +2009-03-30 16:52 +0000 [r185089] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar + 2009) | 45 lines Merged revisions 185031 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar + 2009) | 39 lines Fix queue weight behavior so that calls in + low-weight queues are not inappropriately blocked. (This is + copied and pasted from the review request I made for this patch) + Asterisk has some odd behavior when queue weights are used. The + current logic used when potentially calling a queue member is: If + the member we are going to call is part of another queue and + _that other queue has any callers in it_ and has a higher weight + than the queue we are calling from, then don't try to contact + that member. The issue here is what I have marked with + underscores. If the higher-weighted queue has any callers in it + at all, then the queue member will be unreachable from the + lower-weighted queue. This has the potential to be really really + bad if using a queue strategy, such as leastrecent or + fewestcalls, with the potential to call the same member + repeatedly. The fix proposed by garychen on issue 13220 is very + simple and, as far as I can see, works well for this situation. + With this set of changes, the logic used becomes: If the member + we are going to call is part of another queue, the other queue + has a higher weight than the queue we are calling from, and the + higher weight queue has at least as many callers as available + members, then do not try to contact the queue member. If the + higher weighted queue has fewer callers than available members, + then there is no reason to deny the call to this member since the + other queue can afford to spare a member. Since the fix involved + writing a generic function for determining the number of + available members in the queue, I also modified the is_our_turn + function to make use of the new num_available_members function to + determine if it is our turn to try calling a member. There is one + small behavior change. Before writing this patch, if you had + autofill disabled, then if you were the head caller in a queue, + you would automatically be told that it was your turn to try + calling a member. This did not take into account whether there + were actually any queue members available to take the call. Now + we actually make sure there is at least one member available to + take the call if autofill is disabled. (closes issue #13220) + Reported by: garychen Review: + http://reviewboard.digium.com/r/202/ ........ ................ + +2009-03-30 14:43 +0000 [r184951] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | + 21 lines Merged revisions 184947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | + 14 lines Improve our handling of T38 in the initial INVITE from a + device. We now answer with matching media streams to what is + requested. If an INVITE is received with both a T38 and RTP media + stream this means we answer with both. For any outgoing calls + created as a result of this inbound one no T38 is requested in + the initial INVITE. Instead if we start receiving udptl packets + we trigger a reinvite on the outbound side. (closes issue #12437) + Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu + Review: http://reviewboard.digium.com/r/208/ ........ + ................ + +2009-03-30 13:57 +0000 [r184913] Russell Bryant <russell@digium.com> + + * channels/h323/Makefile.in, /: Merged revisions 184910 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 + Mar 2009) | 4 lines Fix build error when chan_h323 is not being + built. (reported by cai1982 in #asterisk-dev) ........ + +2009-03-29 05:56 +0000 [r184839-184846] Russell Bryant <russell@digium.com> + + * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) + | 13 lines Merged revisions 184842 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) + | 5 lines Ensure targs variable is fully initialized. (closes + issue #14758) Reported by: tim_ringenbach ........ + ................ + + * channels/Makefile, /: Merged revisions 184838 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | + russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines + Simplify chan_h323 build to not require a second run of "make". + (closes issue #14715) Reported by: jthurman Patches: + h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license + 614) Tested by: tzafrir, russell ........ + +2009-03-27 19:21 +0000 [r184779] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c, main/timing.c, main/channel.c, /, + bridges/bridge_softmix.c, include/asterisk/timing.h, + include/asterisk/channel.h: Merged revisions 184762 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar + 2009) | 12 lines Improve timing interface to remember which + provider provided a timer The ability to load/unload timing + interfaces is nice, but it means that when a timer is allocated, + it may come from provider A, but later provider B becomes the + 'preferred' provider. If this happens, all timer API calls on the + timer that was provided by provider A will actually be handed to + provider B, which will say WTF and return an error. This patch + changes the timer API to include a pointer to the provider of the + timer handle so that future operations on the timer will be + forwarded to the proper provider. (closes issue #14697) Reported + by: moy Review: http://reviewboard.digium.com/r/211/ ........ + +2009-03-27 18:12 +0000 [r184707-184729] Russell Bryant <russell@digium.com> + + * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 + Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure + we use the best RNG available. ........ + + * apps/app_queue.c, apps/app_voicemail.c, main/cli.c, + include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c: + Merged revisions 184693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 | + russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines + Change global_app_buf to ast_str_thread_global_buf. ........ + +2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp <jcolp@digium.com> + + * /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7 + lines Fix a potential timer leak in bridge_softmix. It is + possible for a bridge to be created without actually being used. + In that scenario a timing file descriptor would be opened and not + closed. To fix this the timing file descriptor is now closed in + the destroy callback, not the thread function. ........ + + * /, res/res_agi.c: Merged revisions 184673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | + file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix + speech structure leak in the AGI speech recognition integration. + The AGI dialplan applications did not destroy the speech + structure automatically if it was not destroyed by the running + AGI script. They will now do this. (issue LUMENVOX-15) ........ + + * /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2 + lines Remove a cast that is not needed. ........ + +2009-03-27 14:09 +0000 [r184632] Russell Bryant <russell@digium.com> + + * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, + res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions + 184630 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | + russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines + Change g_eid to ast_eid_default. ........ + +2009-03-27 13:59 +0000 [r184612-184629] Joshua Colp <jcolp@digium.com> + + * /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6 + lines Fix a potential race condition when creating a software + based mixing bridge. It was possible for no timer to become + available between creating the bridge and starting it. We now + open a timer when creating it and keep it open until the bridge + is destroyed. ........ + + * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | + 16 lines Merged revisions 184565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 + lines Fix an issue where nat=yes would not always take effect for + the RTP session on outgoing calls. If calls were placed using an + IP address or hostname the global nat setting was copied over but + was not set on the RTP session itself. This caused the RTP stack + to not perform symmetric RTP actions. (closes issue #14546) + Reported by: acunningham ........ ................ + +2009-03-27 02:35 +0000 [r184514-184552] Russell Bryant <russell@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) + | 20 lines Fix some issues with rwlock corruption that caused + deadlock like symptoms. When dvossel and I were doing some load + testing last week, we noticed that we could make Asterisk trunk + lock up instantly when we started generating a bunch of calls. + The backtraces of locked threads were bizarre, and many were + stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a + number of places where a backtrace would be loaded into an + invalid index of the backtrace array. It's an off by one error, + which ends up writing over the rwlock itself. 2) Ensure that in + the array of held locks, we NULL out an index once it is not + being used so that it's not confusing when analyzing its + contents. 3) Remove a bunch of logging referring to an rwlock + operating being done with "deep reentrancy". It is normal for + _many_ threads to hold a read lock on an rwlock. ........ + + * /, main/file.c: Merged revisions 184515 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | + russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines + Don't act surprised if we get a -1 indication. ........ + + * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 + Mar 2009) | 2 lines Pass more useful information through to lock + tracking when DEBUG_THREADS is on. ........ + +2009-03-26 22:19 +0000 [r184454] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile, /: Merged revisions 184448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar + 2009) | 9 lines Merged revisions 184447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar + 2009) | 3 lines use new, improved 8kHz prompts ........ + ................ + +2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant <russell@digium.com> + + * /, main/event.c: Merged revisions 184344 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | + russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines + Remove unneeded AST_LIST_ENTRY() and comment on the purpose of + ast_event_ref. ........ + + * include/asterisk/_private.h, channels/chan_iax2.c, + channels/chan_dahdi.c, include/asterisk/event.h, + apps/app_minivm.c, res/ais/evt.c, main/event.c, + include/asterisk/strings.h, main/asterisk.c, + channels/chan_mgcp.c, apps/app_voicemail.c, + channels/chan_unistim.c, include/asterisk/devicestate.h, /, + channels/chan_sip.c, main/devicestate.c: Merged revisions 184339 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 + Mar 2009) | 35 lines Improve performance of the ast_event cache + functionality. This code comes from + svn/asterisk/team/russell/event_performance/. Here is a summary + of the changes that have been made, in order of both invasiveness + and performance impact, from smallest to largest. 1) Asterisk + 1.6.1 introduces some additional logic to be able to handle + distributed device state. This functionality comes at a cost. One + relatively minor change in this patch is that the extra + processing required for distributed device state is now + completely bypassed if it's not needed. 2) One of the things that + I noticed when profiling this code was that a _lot_ of time was + spent doing string comparisons. I changed the way strings are + represented in an event to include a hash value at the front. So, + before doing a string comparison, we do an integer comparison on + the hash. 3) Finally, the code that handles the event cache has + been re-written. I tried to do this in a such a way that it had + minimal impact on the API. I did have to change one API call, + though - ast_event_queue_and_cache(). However, the way it works + now is nicer, IMO. Each type of event that can be cached (MWI, + device state) has its own hash table and rules for hashing and + comparing objects. This by far made the biggest impact on + performance. For additional details regarding this code and how + it was tested, please see the review request. (closes issue + #14738) Reported by: russell Review: + http://reviewboard.digium.com/r/205/ ........ + +2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | + file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix + issue with a T38 reinvite being sent even if not configured to do + so. If we receive a T38 request negotiate control frame we should + only attempt to do so if the option is enabled on the dialog. + ........ + + * main/bridging.c, /: Merged revisions 183652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 | + file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix + a minor logic flaw with the bridge generic thread. We only want + to move the channel pointers that are actually present. ........ + +2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons <eliels@gmail.com> + + * main/asterisk.c, /: Merged revisions 184220 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | + 19 lines Merged revisions 184188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | + 13 lines Avoid destroying the CLI line when moving the cursor + backward and trying to autocomplete. When moving the cursor + backward and pressing TAB to autocomplete, a NULL is put in the + line and we are loosing what we have already wrote after the + actual cursor position. (closes issue #14373) Reported by: eliel + Patches: asterisk.c.patch uploaded by eliel (license 64) Tested + by: lmadsen ........ ................ + +2009-03-25 14:40 +0000 [r184150-184221] Russell Bryant <russell@digium.com> + + * main/timing.c, /: Merged revisions 184219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 | + russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines + Include poll-compat.h ........ + + * main/timing.c, /: Merged revisions 184151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 | + russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines + Change poll() to ast_poll(). ........ + + * utils/Makefile, /, include/asterisk/compat.h: Merged revisions + 184147 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | + russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines + Fix build issues on Mac OSX. (closes issue #14714) Reported by: + ygor ........ + +2009-03-24 22:42 +0000 [r184082] Mark Michelson <mmichelson@digium.com> + + * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar + 2009) | 15 lines Merged revisions 184078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar + 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. + The 'digit' variable is guaranteed to be non-NULL, so the if + statement could never evaluate true. Changing to ast_strlen_zero + makes the logic correct. This was found while reviewing + ast_channel_ao2 code review. ........ ................ + +2009-03-24 22:02 +0000 [r184041-184044] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 184043 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 | + russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines + Put siren7 and siren14 in ast_best_codec() just so they're in + there somewhere. ........ + + * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) + | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low + and =medium The default codec configuration for chan_iax2 is + bandwidth=low. I noticed slin16 being negotiated as the codec in + some test calls, but that no longer happens after this change. + ........ + +2009-03-24 15:29 +0000 [r183868-183917] Tilghman Lesher <tlesher@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 183914 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 + (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) + | 3 lines Additionally note that the operator option needs an 'o' + extension. (Related to issue #14731) ........ ................ + + * /, main/http.c: Merged revisions 183865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | + tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines + Allow browsers to cache images and other static content. (This is + a regression over 1.4) ........ + +2009-03-23 19:00 +0000 [r183769] Mark Michelson <mmichelson@digium.com> + + * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar + 2009) | 13 lines Merged revisions 183700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar + 2009) | 7 lines Fix a memory leak in res_monitor.c The only way + that this leak would occur is if Monitor were started using the + Manager interface and no File: header were given. Discovered + while reviewing the ast_channel_ao2 review request. ........ + ................ + +2009-03-23 18:12 +0000 [r183704] Leif Madsen <lmadsen@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) + | 7 lines Fixes a documentation error introduced during the CLI + cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: + ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) + Tested by: lmadsen ........ + +2009-03-20 17:09 +0000 [r183564] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183560 | russell | 2009-03-20 12:00:58 -0500 + (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) + | 2 lines Fix a crash in IAX2 registration handling found during + load testing with dvossel. ........ ................ + +2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons <eliels@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | + 2 lines Remove duplicate <description> inside the xml + documentation. ........ + +2009-03-19 19:20 +0000 [r183337] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 + (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) + | 8 lines Delay signalling progress until a PRI channel really + signals progress. (closes issue #13034) Reported by: klaus3000 + Patches: 20090316__bug13034.diff.txt uploaded by tilghman + (license 14) patch_trunk_183progress_klaus3000.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000 ........ + ................ + +2009-03-19 18:20 +0000 [r183263] Russell Bryant <russell@digium.com> + + * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Merged revisions 183242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) + | 10 lines Merged revisions 183241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) + | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving + like expected. ........ ................ + +2009-03-19 18:12 +0000 [r183247] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | + mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 + lines Fix a memory leak associated with queues. For every attempt + that app_queue made to place an outbound call to a queue member, + we would allocate a queue_end_bridge structure. When the bridge + for the call had completed, we would free the structure. + Unfortunately not all call attempts actually end up bridged to a + member, so we need to be more selective of when to allocate the + structure. With this change, the allocation occurs in an area + where we can guarantee that the call will be bridged. (closes + issue #14680) Reported by: caspy Patches: 14680.patch uploaded by + mmichelson (license 60) Tested by: caspy ........ + +2009-03-19 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-beta1 + +2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar + 2009) | 20 lines Merged revisions 183115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar + 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls + would erroneously report the device as "in use." A user was + having an issue where if an outgoing SIP call was canceled, the + SIP device would remain in use if we had not received any + response to the initial INVITE we sent out. The SIP device would + remain in use until the autocongestion timer was exhausted. I + tracked down the cause of this to be the section of code I am + removing here. I asked several people what the purpose of this + code was meant to be, but no one could give me any sort of answer + as to why this was here. The person who was having this issue has + been using this patch for several months and it has stopped the + problems they have had. AST-196 ........ ................ + +2009-03-19 15:45 +0000 [r183068-183111] 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 [r183031] 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 14:39 +0000 [r182947] 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 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/chan_h323.h, 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: 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 16:46 +0000 [r182592] 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 15:01 +0000 [r182528-182534] 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 05:54 +0000 [r182453] 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 ........ ................ + +2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) + | 8 lines OPENR2 uses an incorrect string value if the extension + delimiter is not present. * Fixed OPENR2 using an incorrect + string value if the extension delimiter is not present in the + Dial() function. This was fixed for SS7 and PRI in trunk + -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, + PRI, and others. * Removed trailing whitespace that appeared with + OPENR2. ........ + +2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com> + + * /: svnmerge init + + * / (added): Create a branch for 1.6.2 + +2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com> + + * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + configure, include/asterisk/autoconfig.h.in, configure.ac, + CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This + commit introduces official support for R2 signaling in + chan_dahdi. The modifications to chan_dahdi, and the supporting + library, LibOpenR2, were both written by Moises Silva. Many users + are using this code, or a variant of it, in Asterisk 1.2, 1.4 and + 1.6 in Brazil, México and Argentina. An unknown number of users + (but at least 1) are using it in each of the following countries: + Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. + To use this code, LibOpenR2 must be installed from + http://www.libopenr2.org/. Information about configuration can be + found in configs/chan_dahdi.conf.sample. The code committed is + the most up to date version, which was being maintained in + svn/asterisk/team/moy/mfcr2/. I would also like to include a + Thank You to the many others that tested this code beyond those + listed in this commit message. These are the names that I could + find in the mantis issue. (closes issue #12509) Reported by: moy + Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested + by: moy, korihor, viniciusfontes, Skarmeth, loloski, + asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, + ecarruda, rtorresduque, PTorres, ychen Review: + http://reviewboard.digium.com/r/40/ + +2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: 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 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_env.c: 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 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 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com> + + * main/channel.c: 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 [r182029-182121] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Change faulty comparison used when announcing + average hold minutes and seconds (closes issue #14227) Reported + by: caspy + + * main/features.c: Remove ast_ prefix from functions which are not + public. + + * /, main/features.c: 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 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: 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) + +2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: improve a bit of suboptimal code + +2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com> + + * /: Merged revisions 181898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 Just + recording the v1.4 change in trunk since it originally came from + here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500 + (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated + property when generating the version string. Copied the + make_version file from Asterisk trunk. ........ + +2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Run the macro on the queue member's channel + when he answers, not the caller's channel. + + * /, channels/chan_sip.c: 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 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com> + + * main/translate.c: 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 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: 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 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 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: Fix crash when sleep and retries argument was + not given to RetryDial application. (closes issue #14647) + Reported by: sherpya + +2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com> + + * build_tools/make_version: Whitespace chages. + + * build_tools/make_version: Use the correct branch integrated + property when generating the version string + +2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info> + + * configs/sip.conf.sample: Provide correct hint to debug SIP + trouble in the default config (closes issue #14646) Reported by: + strk + +2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com> + + * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable + thread-safe. + +2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac: 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 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com> + + * main/channel.c: Make handling of the BRIDGEPVTCALLID variable + thread-safe. + + * main/channel.c, /: 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 17:34 +0000 [r181371] David Vossel <dvossel@digium.com> + + * channels/iax2-parser.h, /, channels/chan_iax2.c: 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 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: 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) ........ + +2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2, + since it apparently doesn't work too well during startup. + +2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: 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 16:19 +0000 [r181292] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt: Replace contents of this doc with a + pointer to its new home + +2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix segfault when dialing a typo'd queue If + trying to dial a non-existent queue, there would be a segfault + when attempting to access q->weight, even though q was NULL. This + problem was introduced during the queue-reset merge and thus only + affects trunk. (closes issue #14643) Reported by: alecdavis + +2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com> + + * apps/app_confbridge.c: Don't play the "you are about to be placed + into the conference" and "the leader has left the conference" + sounds if the quiet option is enabled. (reported by Vadim Lebedev + on the asterisk-dev list) + +2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com> + + * utils/Makefile, include/asterisk/utils.h, + include/asterisk/astmm.h, channels/chan_sip.c, + channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c, + pbx/pbx_config.c: 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 02:25 +0000 [r181099] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt: tabs to spaces + +2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Add missing comment that quotes RFC 3891 + + * /, channels/chan_sip.c: 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. ........ + +2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com> + + * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h, + main/heap.c, include/asterisk/strings.h, + include/asterisk/hashtab.h, main/astobj2.c, + include/asterisk/heap.h: Add MALLOC_DEBUG to various utility + APIs, so that memory leaks can be tracked back to their source. + (related to issue #14636) + + * main/pbx.c: Spacing changes only + +2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac, autoconf/ast_prog_sed.m4, + autoconf/ast_check_gnu_make.m4: 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 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt: Add some notes on getting in + contact with the dev community + + * doc/google-soc2009-ideas.txt: Remove difficulty and language + specifiers + + * doc/google-soc2009-ideas.txt: Expand upon documentation of + manager event project + +2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info> + + * CHANGES: list the move of the astvarrundir from /var/run to + /var/run/asterisk (actually its $(localstatedir)/run/asterisk + Makes setups with asterisk as non-root easier to manage because + you can setup permissions on this dir instead of touching a file + and setting permissions on that. Files that come to mind are + asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell + +2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt: add more projects + + * doc/google-soc2009-ideas.txt: add more project ideas + +2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com> + + * main/manager.c: Reset the thread local string buffer when + handling the UserEvent action. (closes issue #14593) Reported by: + JimDickenson + +2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt: Add current mentors list, and first + pass on a project list broken out of "PineMango" I will work on + adding projects that have been sent to be via email tomorrow. + +2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com> + + * include/asterisk/rtp.h, include/asterisk/extconf.h, + main/devicestate.c, include/asterisk/tcptls.h, main/enum.c, + include/asterisk/callerid.h, include/asterisk/doxyref.h, + include/asterisk/event.h, include/asterisk/audiohook.h, + include/asterisk/dsp.h, include/asterisk/timing.h, + include/asterisk/udptl.h, include/asterisk/dlinkedlists.h, + include/asterisk/utils.h, include/asterisk/devicestate.h, + include/asterisk/taskprocessor.h, include/asterisk/enum.h, + include/asterisk/astobj2.h, include/asterisk/config.h, + include/asterisk/channel.h, include/asterisk/manager.h, + include/asterisk/heap.h, include/asterisk/logger.h, + include/asterisk/http.h, include/asterisk/res_odbc.h, + include/asterisk/app.h, main/tcptls.c, + include/asterisk/linkedlists.h, include/asterisk/sched.h, + include/asterisk/datastore.h, include/asterisk/lock.h, + include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen + documentation for API changes from 1.6.0 to 1.6.1 Copied from my + review board description: This is a continuation of the API + changes documentation started for describing changes between + releases. Most of the API changes were pretty simple needing only + to be brought to attention via the new "Asterisk API Changes" + list. However, if you see anything that needs further explanation + feel free to supplement what is there. The current method of + documenting is to add (in the header file): \version <ver number> + <description of changes> and then to add the function to the + change list in doxyref.h on the AstAPIChanges page. I also made + sure all the functions that were newly added were tagged with + \since 1.6.1. I think this is a good habit to start both for the + historical aspect as well as for the future ability to easily add + a "New Asterisk API" page. Review: + http://reviewboard.digium.com/r/190/ + +2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com> + + * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas + list + +2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com> + + * contrib/asterisk-ng-doxygen: Make some minor updates to the + doxygen configuration - add bridges directory to be processed - + add some res/ subdirs - alphabetize subdirs - use consistent + indentation + +2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: 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 17:26 +0000 [r180534] David Vossel <dvossel@digium.com> + + * /, main/enum.c: 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 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: 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 ........ + + * /, configs/voicemail.conf.sample, apps/app_voicemail.c: 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 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info> + + * Makefile: Make sure we terminate the first s| command so we can + actually produce correct files. + +2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com> + + * main/frame.c, /, include/asterisk/frame.h, main/rtp.c: 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/ + ........ + +2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com> + + * channels/chan_bridge.c (added), main/Makefile, + bridges/bridge_simple.c, bridges/bridge_softmix.c, + include/asterisk/channel.h, bridges/bridge_multiplexed.c, + CHANGES, Makefile, include/asterisk/bridging_technology.h + (added), bridges (added), bridges/bridge_builtin_features.c, + include/asterisk/bridging_features.h (added), + include/asterisk/bridging.h (added), apps/app_confbridge.c + (added), main/bridging.c (added), bridges/Makefile: Merge phase 1 + support for the new bridging architecture. This commit brings in + the bridging core, bridging technologies, and the ConfBridge + application. For usage information on the ConfBridge application + please see the output of "core show application ConfBridge" from + the CLI. For API documentation please see the doxygen page + describing the architecture and the documentation for each API + call. Review: http://reviewboard.digium.com/r/93/ + +2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com> + + * contrib/editors/asterisk.vim: Also highlight the preamble and + postamble + + * contrib/editors/ael.vim (added), contrib/editors/asterisk.vim + (added), contrib/editors (added), contrib/editors/asteriskvm.vim + (added): Add syntax coloring files for Vim, including a new one + for AEL + +2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Resolve object matching issues related to + the removal of the sip_user object. Previously, chan_sip had both + sip_peer and sip_user objects in memory. A patch went in to + remove sip_user to simplify the code, since everything could be + done with just sip_peer. This patch resolves some regressions + found that were introduced by those changes. This code comes from + svn/asterisk/team/group/sip-object-matching/. Here is a list of + the changes that have been made: 1) When doing a match by name + with the find_peer() function, make it much easier to specify + which objects should be matched by having a parameter that + specifies exactly which object types should be considered. Also, + update find_by_name() to handle this parameter. Finally, update + all code to use the new option values. 2) When looking up an + object for an outbound request by name, consider peers only. + (create_addr()) 3) Only match peers on an incoming registration + request. 4) When doing authentication (except for SUBSCRIBE), + look up users by name, instead of all objects by name. 5) When + doing authentication (except for SUBSCRIBE), after looking for a + user by name, look for a peer by IP address, instead of all + objects by IP address. 6) When handling the SIP qualify CLI + command or manager action, look for a peer by name, instead of + any object by name. 7) When handling the SIP unregister CLI + command, look for a peer by name, instead of any object by name. + 9) In sip_do_debug_peer(), search for a peer by name, instead of + any object by name. 9) When handling the SIPPEER() dialplan + function, search for a peer by name, instead of any object by + name. 10) In the following session timer related functions, + st_get_se(), st_get_refresher(), and st_get_mode(), when looking + for an object for a given sip_pvt using pvt->peername, look for a + peer by name, instead of any object by name. 11) Fix build_peer() + to properly handle the case where separate type=peer and + type=user entries were specified in sip.conf. (closes issue + #14505) Reported by: lmadsen Review: + http://reviewboard.digium.com/r/172/ + +2009-03-04 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com> + + * main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c, + main/alaw.c: Spacing changes only + +2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com> + + * /, main/callerid.c: 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) ........ + +2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic" + pickups to work when we wish to ignore the context When the + subscription context for a call pickup subscription differs from + the context of the call pickup target, there's not an easy way to + divine what context should be used for the pickup. The way to + work around this is to use PICKUPMARK as the context for the + pickup. This has been documented in the sip.conf.sample file + (ABE-1708) closes issue #14567 submitted by: alecdavis + +2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: 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 [r180079] Steve Murphy <murf@digium.com> + + * utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by + mistake. Already done to 1.6.x + +2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com> + + * main/channel.c, include/asterisk/app.h, apps/app_read.c, + main/app.c: 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 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com> + + * /, configs/queues.conf.sample, apps/app_queue.c: 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 ........ + +2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com> + + * utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, + main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl, + main/ast_expr2.c: 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 22:01 +0000 [r179972] David Vossel <dvossel@digium.com> + + * apps/app_meetme.c: app_meetme not setting filename and fileformat + correctly for realtime When app_meetme finds a realtime + conference, it doesn't get the filename and fileformat correctly + when 'r' is set. Now app_meetme first checks to see if fileformat + and filename are declared in the db, if they're not it checks the + .conf file, if its not declared there either it then uses + defaults. (closes issue #14545) Reported by: dalbaech Patches: + app_meetme-realtime5.patch uploaded by dvossel (license 671) + Realtime_Conference_Record_workaround.txt uploaded by dalbaech + (license 705) Tested by: dvossel, dalbaech Review: + http://reviewboard.digium.com/r/180/ + +2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com> + + * res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation + for timing modules used in Asterisk This document specifies the + timing modules available in Asterisk beginning with Asterisk + 1.6.1. The document goes into detail about the differences + between each and gives a general overview of what timing is used + for in Asterisk. There is also a section which can be used to + help customize your setup or to troubleshoot timing issues you + may have. I also added messages to the DAHDI timing test used in + res_timing_dahdi.c that points to this new documentation if + people experience problems. Big thanks to all who contributed + comments on this. (closes issue #14490) Reported by: mmichelson + Patches: timing.txt uploaded by mmichelson (license 60) Review: + http://reviewboard.digium.com/r/164/ + +2009-03-03 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com> + + * apps/app_directed_pickup.c: fix a leaked channel lock (and future + deadlock) when we try to pick up our own channel + +2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: 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 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com> + + * pbx/pbx_spool.c: Convert pbx_spool to use string fields instead + of statically-sized buffers. In tests run after making this + conversion, I noticed an approximate 85% reduction in memory + usage for call file processing. Review: + http://reviewboard.digium.com/r/168/ + +2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com> + + * main/channel.c, /: 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 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Please prefix default values with DEFAULT + +2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: 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 13:54 +0000 [r179609] Russell Bryant <russell@digium.com> + + * main/channel.c, /: 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 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: 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 23:36 +0000 [r179533] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: 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 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: 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 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com> + + * res/res_timing_timerfd.c: Fix a reference leak in + timerfd_set_rate(). (found during a debugging session with + dvossel and mmichelson.) + + * main/channel.c, /: 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 20:16 +0000 [r179396] Jason Parker <jparker@digium.com> + + * /, main/editline/configure, main/editline/np/unvis.c, + main/editline/sys.h, main/editline/configure.in: 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 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe + if db is not loaded) + +2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Do not try to remove a registration + scheduled item if the scheduler context has already been + destroyed. (closes issue #14580) Reported by: alecdavis + + * main/audiohook.c: Fix issue where changing the volume of both + directions of audio did not work. (closes issue #14574) Reported + by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK + (license 545) + +2009-03-01 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com> + + * apps/app_speech_utils.c: Swap reversed timevals. This was pointed + out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! + + * channels/chan_sip.c: 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_ERROR was returned by __sip_xmit. When + retrans_pkt was called, this inevitably resulted in the reading + and writing of freed memory. XMIT_ERROR 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 21:47 +0000 [r179164] Russell Bryant <russell@digium.com> + + * res/res_ais.c, doc/distributed_devstate.txt, + configs/ais.conf.sample: Mark res_ais as experimental, as the + binary event format is subject to change. + +2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c: If config file is blank, don't load + module. (Closes issue #14563) + +2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com> + + * UPGRADE.txt: Add a note about the ordering of entries in sip.conf + in 1.6.1. + +2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: Add reload support to chan_skinny. + Special thanks goes to DEA who had to redo this patch twice + because we first put unload/load support in and later redid the + way we configure devices and lines. (closes issue #10297) + Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by + wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA + (license 3) With mods by me based on feedback from wedhorn and + Russell and seanbright Tested by: DEA, mvanbaak, pj Review: + http://reviewboard.digium.com/r/130/ + +2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com> + + * doc/tex/channelvariables.tex: 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 15:51 +0000 [r179021] Russell Bryant <russell@digium.com> + + * sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound + packages. In passing, also fix downloading SLIN16 extra sound + packages. (closes issue #14565) Reported by: jtodd + +2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com> + + * /, main/features.c, configs/features.conf.sample: 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 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com> + + * main/features.c, CHANGES, configs/features.conf.sample: Sound + confirmation of call pickup success. (closes issue #13826) + Reported by: azielke Patches: pickupsound2-trunk.patch uploaded + by azielke (license 548) __20081124_bug_13826_updated.patch + uploaded by lmadsen (license 10) Tested by: lmadsen + +2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last + fix A return statement was missing which caused unexpected cli + output. issue #14479 + +2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com> + + * apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent + compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to + break my dev-mode build. Not a problem in 1.6.x. + + * /, main/features.c: 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 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com> + + * main/file.c: Fix an issue where the timer for file playback would + not be stopped if DAHDI was not installed. (closes issue #14541) + Reported by: grant + +2009-02-26 15:50 +0000 [r178767] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: 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/ + +2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com> + + * main/indications.c: Ensure there is a valid tone part before + trying to play tones. (closes issue #14558) Reported by: + alecdavis + +2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net> + + * configs/res_snmp.conf.sample: Clarifications on the different + models and reference to further docs. + +2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com> + + * README: another minor commit to test post-commit script changes + (now testing post-revprop-change as well, third try) + + * README: minor commit to test post-commit script changes + +2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com> + + * main/stdtime/localtime.c: Picky, picky buildbots + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/stdtime/localtime.c: Use notification when timezone files + change and re-scan then. (closes issue #14300) Reported by: + jamessan Patches: 20090127__bug14300.diff.txt uploaded by + tilghman (license 14) 20090224__bug14300.diff uploaded by + jamessan (license 246) Tested by: jamessan Review: + http://reviewboard.digium.com/r/136/ + + * res/res_odbc.c: Oops, wrong direction of command + +2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com> + + * /, main/asterisk.c: 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 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com> + + * /, configs/extensions.conf.sample: 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: Apparently, a void cast doesn't override + warn_unused_result. + + * main/asterisk.c: The 3 possible errors with pipe(2) are all + impossible in this situation. + +2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com> + + * /, main/rtp.c: 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 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com> + + * utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the + process, instead of depending upon the astcanary process being + inherited by init. + + * utils/astcanary.c: 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 17:42 +0000 [r178300] David Vossel <dvossel@digium.com> + + * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows + manager command to see if IAX link is trunked and encrypted. + Displays what kind of encryption is enabled as well. Manager + command "iaxpeers" now shows if a link is trunked and encrypted. + Instead of encryption saying simply "yes" or "no", it now + displays what type of encryption is enabled and if keyrotation is + on or not. (closes issue #14427) Reported by: snuffy Patches: + iax_show_trunks.diff uploaded by snuffy (license 35) + 2009022200_iax2_show_trunkencryption.diff.txt uploaded by + mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: + http://reviewboard.digium.com/r/173/ + +2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: 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 23:11 +0000 [r178142] Russell Bryant <russell@digium.com> + + * /, main/rtp.c: 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 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com> + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Permit emailsubject and emailbody to be set per mailbox. (closes + issue #14372) Reported by: fhackenberger Patches: + voicemail_individual_subject_and_body_1.6.1 uploaded by + fhackenberger (license 592) with additional fixes by Corydon76 + (license 14) + +2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: update the new manager commands in + chan_skinny to match chan_sip's headers. requested by oej. + +2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Changes the way keyrotation is enabled by + default Key rotation was enabled by default by setting the global + encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this + is that if encryption is not enabled, and the encryption method + is set to anything except 0, the peer appears to have encryption + enabled when issuing a "iax2 show peers". Rather than have the + key rotation bit always set by default, it is now only set when + an encryption method is enabled. (closes issue #14523) Reported + by: mvanbaak + +2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info> + + * CHANGES: list the addition of the SKINNY manager actions in the + CHANGES file. + +2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com> + + * tests/test_sched.c, main/sched.c: Fix a regression in scheduler + entry ordering, and add a regression test for it. (closes issue + #14522) Reported by: pj Tested by: russell + +2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info> + + * doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of + manager commands to chan_skinny Added: SKINNYdevices + SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521) + Reported by: mvanbaak Review: + http://reviewboard.digium.com/r/170/ + +2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: On update, test against the existence of + sipregs. + +2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info> + + * main/asterisk.c: add extra check for sysinfo/sysctl (closes issue + #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff + uploaded by snuffy (license 35) + +2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com> + + * main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace, + minor coding guideline fixes, and start beefing up the hashtab + documentation a bit. + +2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com> + + * include/asterisk/indications.h: Fix build issues on Solaris and + OpenBSD. (closes issue #14512) Reported by: snuffy + +2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info> + + * Makefile, contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.archlinux.asterisk, + contrib/scripts/safe_asterisk: set + ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When + running asterisk as non-root and without this patch the pidfile + wants to go into /var/run/asterisk.pid. This directory is not + writable for the non-root user and changing permissions is not an + option. Putting it in /var/run/asterisk/asterisk.pid makes it + possible to set permissions on the /var/run/asterisk dir so + everything works as it should be. Patched committed is based on + pabelanger's patch. (closes issue #13153) Reported by: pabelanger + Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by + mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/ + + * channels/chan_sip.c: make chan_sip.c compile on OpenBSD again. + +2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 177786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) + | 9 lines Don't print the CR-NL combination when we aren't + outputting to the manager. An embedded CR-NL in a CLI command + screws up several AMI parsers that don't expect to see that + combination in the middle of output. (Closes issue #14305) + Reported by: martins Patch by: tilghman ........ + + * /, include/asterisk/threadstorage.h: Merged revisions 177701 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) + | 3 lines This exception does not appear to still be true for + Solaris 10, and OpenSolaris definitely needs it to be removed. + Fixed for snuff-home on -dev channel. ........ + +2009-02-20 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4 + Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a + pages_transferred integer to indicate the number of pages + transferred (so far) during the fax session. The + spandsp-0.0.6pre4 release removed the pages_transferred integer + and replaced it with two different integers - pages_tx and + pages_rx. This revision uses the new integers for + spandsp-0.0.6pre4 while maintaining backwards compatibility for + previous spandsp releases. + +2009-02-20 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/app.h, main/app.c, apps/app_system.c: Allow + semicolons to be escaped, when passing arguments to the System + command. (closes issue #14231) Reported by: jcovert Patches: + 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) + corrected_20090113__bug14231__2.diff.txt uploaded by jcovert + (license 551) Tested by: jcovert + + * apps/app_voicemail.c: Oops, merge broke trunk + +2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: Set sip_request ast_str data to NULL so + ast_str_copy allocates space properly in copy_request (issue + #14478) Reported by: erik_dedecker + +2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com> + + * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177540 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was + already pretty 8-bit clean; but I'm still removing the --full + from the flex command so everything is uniform. ........ r177540 + | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines + This patch fixes a problem with 8-bit input to the ast_expr2 + scanner. The real culprit was the --full argument to flex in the + Makefile! This causes a 7-bit scanner to be generated. I reviewed + the rules and found one rule where I needed to specifically + include 8-bit chars for a token. I tested against the text + supplied by ibercom, and all looks very well. This has been there + a surprisingly long time! (closes issue #14498) Reported by: + ibercom Patches: 14498.patch uploaded by murf (license 17) Tested + by: murf ........ + +2009-02-19 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 + Feb 2009) | 7 lines Fix up potential crashes, by reducing the + sharing between interactive and non-interactive threads. (closes + issue #14253) Reported by: Skavin Patches: + 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) + Tested by: Skavin ........ + + * doc/database_transactions.txt (added): Document how to use + database transactions + +2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com> + + * include/asterisk/channel.h: Fix another merge error from 176708 + +2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com> + + * /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb + 2009) | 3 lines If we are able to create a speech structure unset + the ERROR variable in case it was previously set. (issue + #LUMENVOX-13) ........ + +2009-02-19 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: Fix mismerge from revision 176708 pointed out by + Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! + +2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES, + res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction + support + +2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com> + + * CHANGES: Update CHANGES file to include MWI subscription support + that was added some time ago. + +2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com> + + * main/strings.c: Handle negative length and eliminate a condition + that is always true. + +2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com> + + * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | + 34 lines This patch fixes a regression of sorts that was + introduced in rev 24425. It basically fixes AST-190/ABE-1782. + What was wrong: the user has 6000 extensions in one context; and + then 6000 contexts, one per extension. The parser could only + handle about 4893 of the 6000 extens in the single context. This + was due to the regression I mentioned. To get rid of shift/reduce + conflicts, Luigi set up right-recursive lists for globals, + context elements, switch lists, and statements. Right recursive + lists got rid of the warnings, but instead, they use up a + tremendous amount of stack space when the lists are long. I saw + this a few years back, and resolved not to fix it until someone + complained. That day has arrived! After the changes were made, I + ran the regression test suite, and there were no problems. I took + the test case the user provided, and added 100,000 extensions to + the single context, that already had 6,000 extens in it. (I'll + see your 6, and raise you 100!) It takes a few minutes to read it + all in, check it and generate code for it, but no problems. So, I + think I can say that fundamentally, there are no longer any + limits on the number of items you can place in contexts, + statement blocks, switches, or globals, beyond your virt mem + constraints. ........ + +2009-02-18 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com> + + * main/frame.c: fix two very minor bugs: if anyone ever uses + SLINEAR16 as a format in RTP, ensure that the samples are + byte-swapped to network order if needed. also, when a smoother is + operating on a format that has a sample rate other than 8000 + samples per second, use the proper sample rate for computing + delivery timestamps. + +2009-02-18 22:51 +0000 [r177226] David Vossel <dvossel@digium.com> + + * main/features.c: Locking issue in action_bridge and bridge_exec + action_bridge() and bridge_exec() both search for the channels to + bridge to, and then immediately drop the lock. Instead, they + should hold the lock until the masquerade is complete. This will + guarantee the channel remains and prevent any other weirdness + from occurring. In action_bridge() some more weirdness comes into + play. Both channels are needlessly locked at the same time and + perform the exact same logic. It makes sense from a coding + organizational standpoint, but could cause a theoretical deadlock + so I split the code up. There is an issue associated with this, + but since its a rather complicated thing to reproduce I'm not + certain this alone will close it. issue# 14296 Review: + http://reviewboard.digium.com/r/167/ + +2009-02-18 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, + channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx, + channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added), + configure, channels/h323/compat_h323.h, configure.ac, + channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, + channels/h323/ast_h323.h, channels/h323/chan_h323.h, + channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib + as well as the older PWLib Several changes in PTLib have occurred + requiring build time detection. Changes accounted for include the + library name change, config option change, install location + change, and a boolean type change which is handled by + ast_ptlib.h. Also, the sed check has been modified to properly + work with autoconf >= 2.62. (closes issue #14224) Reported by: + bergolth Patches: asterisk-autoconf-sed.patch uploaded by + bergolth (license 661) asterisk-pwlib-v3.patch uploaded by + bergolth (license 661) Tested by: jpeeler + +2009-02-18 19:12 +0000 [r177101] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev + 62297. Enabling this option by default proved to be a bad idea, + as the talker detection is not very reliable. So, make it + optional again, and off by default. (issue #13801) Reported by: + justdave + +2009-02-18 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/config.h: Merged revisions 177096 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) + | 2 lines Document the return value of the update method (as + requested on -dev list) ........ + +2009-02-18 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com> + + * main/utils.c: Fixed error where a check for an zero length, + terminated string was needed. + +2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix ordering of output for a ChannelUpdate + manager event. (closes issue #14497) Reported by: vinsik Patches: + chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) + +2009-02-18 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com> + + * main/utils.c: Need to take into account the \0 terminator of the + old string to determine the amount available. + +2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com> + + * main/pbx.c: This patch fixes merge_contexts_and_delete so it does + not deadlock when hints are present. Reason: when I re-engineered + the merge_and_delete func to reduce its lock time, I failed to + notice that the functions it calls still also do locking as + before. This leads to deadlocks on dialplan reloads, when there + are actually living, subscribed hints registered in the system. + While the reporter come across this problem while using AEL, I + might note that these deadlocks should also happen if + extensions.conf were used. Here I added these routines to pbx.c: + ast_add_extension_nolock add_pri_lockopt + ast_add_extension2_lockopt find_context add_hint_nolock All of + the above routines are static and restricted to be used only + within pbx.c, and more specifically within the + merge_contexts_and_delete routine. They are pretty much the same + as their counterparts except they don't lock contexts or hints. + Most of them now do the real work of their name-alike, with + optional locking via extra arguments, and are called by their + name-alike. The goal was to have the original functions so they + would behave exactly as before. Both PJ and I tested these fixes, + and the deadlocking problem is no longer encountered. (closes + issue #14357) Reported by: pj Patches: 14357.diff uploaded by + murf (license 17) Tested by: pj, murf + +2009-02-18 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com> + + * include/asterisk/heap.h: Add example code for a heap traversal. + + * main/pbx.c: Fix a number of incorrect uses of strncpy(). The big + problem here is that the 3rd argument provided in these uses of + strncpy() did not reserve a byte for the null terminator, leaving + the potential for writing one byte past the end of the buffer. + Aside from this, there were coding guidelines violations with + regards to spacing, as well as hard coded lengths being used + instead of sizeof(). + +2009-02-18 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * channels/chan_sip.c: T38 faxdetect should jump to the 'fax' + extension for incoming calls only The previous implementation of + T38 faxdetect resulted in both sides of the call jumping to a fax + extension when both sides had 't38pt_udptl=yes' and + 'faxdetect=yes' in sip.conf and a 'fax' extension in the current + context. This revision will jump to a 'fax' extension on incoming + calls only. + +2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com> + + * main/rtp.c: suppress smoothers for Siren codecs as well as Speex + and G.723.1 + +2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com> + + * apps/app_milliwatt.c: Remove a dependency that no longer exists. + +2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice + with G723. This commit brings in the changes that were living out + on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder + branch. codec_dahdi.c now always uses signed linear as the simple + codec so that a soft g729 codec will not end up being preferred + to the hardware codec. There are also changes to allow + codec_dahdi.c to feed packets to the hardware in the native + sample size of the codec. This solves problems with choppy audio + when using G723. + +2009-02-17 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /, main/features.c, include/asterisk/channel.h: + Merged revisions 176701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) + | 17 lines Modify bridging to properly evaluate DTMF after first + warning is played The main problem is currently if the Dial flag + L is used with a warning sound, DTMF is not evaluated after the + first warning sound. To fix this, a flag has been added in + ast_generic_bridge for playing the warning which ensures that if + a scheduled warning is missed, multiple warrnings are not played + back (due to a feature evaluation or waiting for digits). + ast_channel_bridge was modified to store the nexteventts in the + ast_bridge_config structure as that information was lost every + time ast_channel_bridge was reentered, causing a hangup due to + incorrect time calculations. (closes issue #14315) Reported by: + tim_ringenbach Reviewed on reviewboard: + http://reviewboard.digium.com/r/163/ ........ + +2009-02-17 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com> + + * tests/test_sched.c: Use constants from inttypes.h to clear up + 32-bit compilation errors + +2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * channels/chan_sip.c: create a UDPTL structure in + create_addr_from_peer() if it does not already exist for T38 This + is required to create a UDPTL structure in + create_addr_from_peer() to handle the scenario where + 't38pt_udptl=yes' is not defined in the [general] section of + sip.conf but is defined the peer's context. I tested this patch + by enabling t38pt_udptl in the [general] section on one system + and only enabling t38pt_udptl in a peer's context on the system + sending a fax. Without the patch, the sending system will fail to + initiate T38 negotiation with the warning message, "No way to add + SDP without an UDPTL structure". When this patch is applied the + sending side will successfully initiate T38 negotiation. + +2009-02-17 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/frame.h: Clear up documentation of + AST_FRIENDLY_OFFSET in frame.h + +2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com> + + * /: Recorded merge of revisions 176661 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009) + | 9 lines Backport change to 1.4: Prior to masquerade, move the + group definitions to the channel performing the masq, so that the + group count lingers past the bridge. (closes issue #14275) + Reported by: kowalma Patches: 20090216__bug14275.diff.txt + uploaded by Corydon76 (license 14) Tested by: kowalma ........ + +2009-02-17 21:22 +0000 [r176666] Russell Bryant <russell@digium.com> + + * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, + res/res_timing_timerfd.c, include/asterisk/timing.h, + main/timing.c: Update the timing API to have better support for + multiple timing interfaces. 1) Add module use count handling so + that timing modules can be unloaded. 2) Implement unload_module() + functions for the timing interface modules. 3) Allow multiple + timing modules to be loaded, and use the one with the highest + priority value. 4) Report which timing module is being use in the + "timing test" CLI command. (closes issue #14489) Reported by: + russell Review: http://reviewboard.digium.com/r/162/ + +2009-02-17 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c: Prior to masquerade, move the group + definitions to the channel performing the masq, so that the group + count lingers past the bridge. (closes issue #14275) Reported by: + kowalma Patches: 20090216__bug14275.diff.txt uploaded by + Corydon76 (license 14) Tested by: kowalma + +2009-02-17 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com> + + * tests/test_sched.c (added), main/sched.c: Significantly improve + scheduler performance under high load. This patch changes the + scheduler to use a max-heap to store pending scheduler entries + instead of a fully sorted doubly linked list. When the number of + entries in the scheduler gets large, this will perform much + better. For much more detailed information on this change, see + the review request. Review: http://reviewboard.digium.com/r/160/ + + * tests/test_heap.c (added): Add a test module for the heap + implementation. Review: http://reviewboard.digium.com/r/160/ + + * main/Makefile, main/heap.c (added), include/asterisk/heap.h + (added): Add an implementation of the heap data structure. A heap + is a convenient data structure for implementing a priority queue. + Code from svn/asterisk/team/russell/heap/. Review: + http://reviewboard.digium.com/r/160/ + +2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net> + + * include/asterisk/config.h: Typo + +2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com> + + * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, + configs/indications.conf.sample, apps/app_playtones.c (added), + include/asterisk/indications.h, apps/app_readexten.c, + apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h, + include/asterisk/_private.h, main/indications.c, main/loader.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, + funcs/func_channel.c, res/snmp/agent.c, main/app.c, + res/res_indications.c (removed), main/asterisk.c: Merge a large + set of updates to the Asterisk indications API. This patch + includes a number of changes to the indications API. The primary + motivation for this work was to improve stability. The object + management in this API was significantly flawed, and a number of + trivial situations could cause crashes. The changes included are: + 1) Remove the module res_indications. This included the critical + functionality that actually loaded the indications configuration. + I have seen many people have Asterisk problems because they + accidentally did not have an indications.conf present and loaded. + Now, this code is in the core, and Asterisk will fail to start + without indications configuration. There was one part of + res_indications, the dialplan applications, which did belong in a + module, and have been moved to a new module, app_playtones. 2) + Object management has been significantly changed. Tone zones are + now managed using astobj2, and it is no longer possible to crash + Asterisk by issuing a reload that destroys tone zones while they + are in use. 3) The API documentation has been filled out. 4) The + API has been updated to follow our naming conventions. 5) Various + bits of code throughout the tree have been updated to account for + the API update. 6) Configuration parsing has been mostly + re-written. 7) "Code cleanup" The code is from + svn/asterisk/team/russell/indications/. Review: + http://reviewboard.digium.com/r/149/ + +2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to + track down a refcount leak. (closes issue #14485) Reported by: + davevg + +2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com> + + * main/pbx.c, apps/app_queue.c: Fix a race condition that caused + device states to become incorrect for hints. The problem here is + that the hint processing code was subscribed to the wrong event + type. So, it started processing state for a hint too soon, before + the device state cache had been updated. Also, fix a similar bug + in app_queue, as it was also subscribed to the wrong event type. + (closes issue #14461) Reported by: alecdavis + +2009-02-17 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net> + + * configs/extconfig.conf.sample: Typo + + * main/config.c: If there are no realtime engines, there's no + reason to check for realtime families + +2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: In this version, we can combine the queries, + because we support dropping nonexistent columns. + + * /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) + | 10 lines After a 'sip reload', qualifies for realtime peers + weren't immediately restarted, instead waiting until the next + registration. We're now caching the qualify across a + reload/restart and starting the qualify immediately upon loading + the peer. (closes issue #14196) Reported by: pdf Patches: + 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) + Tested by: pdf ........ + + * main/strings.c: Might want to update the buffer pointer after a + realloc (or we crash) (closes issue #14485) Reported by: davevg + +2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/sounds.xml: add support for Siren7 and Siren14 flavors of + prompts and music on hold + +2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 + Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not + being relayed correctly during bridging This should have been + committed with rev176247, but I missed it. srcupdate frames no + longer break out of the native bridge, but are not being sent to + the other call leg either. This fixs that. issue #13749 ........ + +2009-02-16 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_skinny.c: Use the correct list macros for deleting + an item from the middle of a list. (issue #13777) Reported by: pj + Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 + (license 14) Tested by: pj + +2009-02-16 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/utils.c, include/asterisk/stringfields.h: Merged + revisions 176216 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb + 2009) | 3 lines fix a flaw in the ast_string_field_build() family + of API calls; these functions made no attempt to reuse the space + already allocated to a field, so every time the field was written + it would allocate new space, leading to what appeared to be a + memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 + -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the + last stringfields commit... don't mark additional space as + allocated if the string was built using already-allocated space + ........ + +2009-02-16 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, + 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it + to be nonblocking atomically Apparently on FreeBSD, attempting to + set the O_NONBLOCKING flag separately from opening the file was + causing an "inappropriate ioctl for device" error. While I cannot + fathom why this would be happening, I certainly am not opposed to + making the code a bit more compact/efficient if it also fixes a + bug. (closes issue #14482) Reported by: ys Patches: meetme.patch + uploaded by ys (license 281) Tested by: ys ........ r176252 | + mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 + lines Remove unused variable and make dev-mode compilation happy + ........ + +2009-02-16 21:30 +0000 [r176248] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Merged revisions 175597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | + dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines + Fixed iax2 key rotation backwards compatibility Turns key + rotation back on by default. Added bit into encryption IE to + indicate whether or not key rotation is supported or not. If it + is not supported then it is not enabled, which insures backwards + compatibility. This eliminates the need for the keyrotate option + in iax.conf, so it has been removed. ........ + +2009-02-16 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com> + + * main/logger.c: Assist proper thread synchronization when stopping + the logger thread. I was finding that on my dev box, occasionally + attempting to "stop now" in trunk would cause Asterisk to hang. I + traced this to the fact that the logger thread was waiting on a + condition which had already been signalled. The logger thread + also need to be sure to check the value of the + close_logger_thread variable. The close_logger_thread variable is + only checked when the list of logmessages is empty. This allows + for the logger thread to print and free any pending messages + before exiting. + +2009-02-16 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c: Can't set debug level 2 (intense + debugging) unless the syntax matches + +2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com> + + * channels/chan_features.c (removed): Remove chan_features. Review: + http://reviewboard.digium.com/r/161/ + +2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 + lines Don't have the Via header stored as a stringfield as it can + change often during the lifetime of a dialog. This issue crept up + with subscriptions on the AA50. When an outgoing NOTIFY is sent a + new branch value is created and the Via header is changed to + reflect it. Since this was a stringfield a new spot in the pool + was used for the value while the old was left untouched/unused. + If the current pool was full a new pool was created. This would + cause memory usage to increase steadily. (issue #AA50-2332) + ........ + +2009-02-16 02:54 +0000 [r175983] Russell Bryant <russell@digium.com> + + * main/channel.c: Make the causes array static, and remove the type + name as it is not needed. + +2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_unistim.c, /, channels/chan_sip.c, + include/asterisk/manager.h, doc/unistim.txt: Merged revisions + 175921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) + | 3 lines fix mis-spelling of the word registered. Reported by + De_Mon on #asterisk-dev. ........ + +2009-02-15 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com> + + * include/asterisk/sched.h, main/sched.c: Make ast_sched_report() + and ast_sched_dump() thread safe. + + * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix + a number of problems with ast_sched_report(). 1) It had numerous + coding guidelines violations with regards to formatting. 2) It + allocated memory using ast_calloc() that was never freed. 3) It + didn't check for failure from the allocation. 4) It used + sprintf() and strcat() to build the result, doing zero checking + to prevent writing past the end of the provided buffer. The + function also lacks API documentation, but that has not been + addressed in this commit. + +2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net> + + * formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb + 2009) | 2 lines format_ilbc does not depend on codec libraries + and can therefore always be made. My mistake. Ursäkta! ........ + + * formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb + 2009) | 2 lines Disable format_ilbc.so by default, like + codec_ilbc.so ........ + + * /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 + lines Make sure that the debug line is not printed on debug level + 0 ........ + +2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com> + + * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset + branch to Asterisk From a user point-of-view, this adds new CLI + commands and Manager Actions to better facilitate the reloading + of queues and the resetting of their statistics. The new CLI + commands are the "queue reload" and "queue reset stats" commands. + The new manager actions are the QueueReload and QueueReset + commands. Review: http://reviewboard.digium.com/r/115 + + * doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for + chanspy starting or stopping (closes issue #14469) Reported by: + caio1982 Patches: chanspy_events2.diff uploaded by caio1982 + (license 22) + +2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com> + + * res/res_jabber.c: fix a few more XML documentation problems + + * main/pbx.c: add missing </para> + +2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c: + Fixed iax2 key rotation backwards compatibility Turns key + rotation back on by default. Added bit into encryption IE to + indicate whether or not key rotation is supported or not. If it + is not supported then it is not enabled, which insures backwards + compatibility. This eliminates the need for the keyrotate option + in iax.conf, so it has been removed. Review: + http://reviewboard.digium.com/r/159/ + +2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, + 13 Feb 2009) | 16 lines Fix a potential crash situation when + using IMAP voicemail If calling into VoiceMailMain when using + IMAP storage, it was possible to crash Asterisk by hanging up the + phone when prompted for a voicemail mailbox. This patch fixes the + issue. While it may appear that this patch is superficial, it + allows code execution to continue to the failure case just below + the IMAP_STORAGE code block where this patch has been applied + (closes issue #14473) Reported by: dwpaul Patches: + voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license + 689) ........ + +2009-02-13 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com> + + * apps/app_record.c: Add an option to keep the recorded file upon + hangup. (closes issue #14341) Reported by: fnordian + +2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com> + + * CHANGES: document G.722.1/.1C support + + * main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h, + channels/chan_h323.c, include/asterisk/frame.h, + formats/format_siren14.c (added), main/rtp.c, + formats/format_siren7.c (added): Add basic (passthrough, + playback, record) support for ITU G.722.1 and G.722.1C (also + known as Siren7 and Siren14) This patch adds passthrough, file + recording and file playback support for the codecs listed above, + with negotiation over SIP/SDP supported. Due to Asterisk's + current limitation of treating a codec/bitrate combination as a + unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are + supported. Along the way, some related work was done: 1) The + rtpPayloadType structure definition, used as a return result for + an API call in rtp.h, was moved from rtp.c to rtp.h so that the + API call was actually usable. The only previous used of the API + all was chan_h323.c, which had a duplicate of the structure + definition instead of doing it the right way. 2) The hardcoded + SDP sample rates for various codecs in chan_sip.c were removed, + in favor of storing these sample rates in rtp.c along with the + codec definitions there. A new API call was added to allow + retrieval of the sample rate for a given codec. 3) Some basic + 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip + *must* decline any media streams offered for these codecs that + are not at the bitrates that we support (otherwise Bad Things + (TM) would result). Review: http://reviewboard.digium.com/r/158/ + +2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * CHANGES: add 'faxbuffers' configuration option information to + CHANGES + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add + dynamic fax buffer configuration option to chan_dahdi.conf When + the 'faxdetect' configuration option is used, one may also want + to use the 'faxbuffers' configuration option in chan_dahdi.conf. + This option will dynamically use the configured 'faxbuffers' + buffer policy on a channel for the life of the call following the + detection of fax tones. The faxbuffers buffer policy will be + reverted during call teardown. An example use of 'faxbuffers' is + below. This example would switch to using 6 buffers with a full + buffer policy. faxbuffers=>6,full + +2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Remove useless string copy, and make sscanf + safe again + +2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds + force encryption option to iax.conf This patch adds + forceencryption=yes as an iax.conf option. When force encryption + is enabled, no unencrypted connections are allowed. This insures + all connections are encrypted. This is a new feature, so CHANGES + and iax.conf.sample are updated as well. (closes issue #13285) + Reported by: sgofferj Tested by: russell Review: + http://reviewboard.digium.com/r/150/ + +2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: 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 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: 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 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Avoid using ast_strdupa() in a loop. + + * build_tools/cflags.xml: 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 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c: correct warning message to not refer + specifically to DAHDI + +2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: 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 17:07 +0000 [r175127] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Setting key rotation to be off by default + Key rotation breaks compatibility between (trunk/1.6.1) and + (1.2/1.4/1.6.0). As a follow up to this, I am investigating + possible ways to allow key rotation to be on by default and not + affect the other branches, but for now it must be turned off. + +2009-02-12 16:57 +0000 [r175125] Russell Bryant <russell@digium.com> + + * /, main/rtp.c: 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 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/astobj2.h, main/astobj2.c: 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 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_gtalk.c: 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 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 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a bit of odd logic for announcing position. + Sync with 1.6.0's logic + + * apps/app_queue.c: 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: 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 + +2009-02-11 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Tell the device state core a change happened when + a channel is freed but not a specific state. We need to do this + because while we know that the freeing of the channel may cause + something to become not in use we do not know this for sure. + There may be another channel that is still up which would cause + it to be in use. (closes issue #13238) Reported by: kowalma + Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 + (license 14) Tested by: alecdavis + +2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c: 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: 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 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Only decrease inringing count if above zero. + (issue #13238) Reported by: kowalma + +2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com> + + * main/slinfactory.c, include/asterisk/slinfactory.h: improve + slinfactory API to remove implicit sample rate and require + explicit sample rate selection by creator of the slinfactory + +2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com> + + * /, main/jitterbuf.c: 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 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Set the type for the peer structure to be a + peer as the default. (closes issue #14447) Reported by: triccyx + + * channels/chan_sip.c: 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 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, apps/app_voicemail.c: Fix0ring build + + * apps/app_stack.c: Remove the usage of the KeepAlive app, as it no + longer exists. + +2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com> + + * apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE + from app_rpt.c. (closes issue #14435) Reported by: D_McNaul + + * apps/app_rpt.c: More intptr_t work. + + * /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 + lines This patch solves some compiler complaints in both 32 and + 64-bit environments. ........ + +2009-02-09 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix something I messed up in the merge I + just did + +2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com> + + * apps/app_externalivr.c: Fixes issue with hangups not being sent + and external process never terminating. The ignore_hangup, + run_dead, and noanswer flags were never initilized to zero + causing hangups to never be issued. If the external script + expects to be notified of a hangup and never receives one, it + runs indefinitely. (closes issue #14251) Reported by: chris-mac + Tested by: dvossel + +2009-02-09 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: 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 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: 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 16:16 +0000 [r174149] Russell Bryant <russell@digium.com> + + * /, res/snmp/agent.c: 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 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * /, channels/chan_sip.c: 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 20:12 +0000 [r174046] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds + immediate yes/no option to iax.conf This is very similar to the + DAHDI immediate=yes option. When the phone is picked up, instead + of giving a dialtone it connects directly to the "s" extension. + Changes where implemented in chan_iax2.c to directly connect to + the "s" extension in the appropriate context when this option is + enabled. Examples explaining its use are added to + iax2.conf.sample. CHANGES has been updated as well. (closes issue + #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk + uploaded by jcovert (license 551) iax.conf.sample.patch uploaded + by jcovert (license 551) Tested by: jcovert, dvossel Review: + http://reviewboard.digium.com/r/143/ + +2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com> + + * channels/chan_dahdi.c: 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 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 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: 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 ........ + +2009-02-06 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com> + + * main/audiohook.c, apps/app_chanspy.c: Always detach and destroy + the whisper and barge audiohooks. Additionally also allow an + audiohook to be detached if it has not been attached. (closes + issue #14414) Reported by: bluecrow76 + +2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com> + + * include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add + a common implementation of a scheduler context with a dedicated + thread. This commit expands the Asterisk scheduler API to include + a common implementation of a scheduler context being processed by + a dedicated thread. chan_iax2 has been updated to use this new + code. Also, as a result, this resolves some race conditions + related to the previous chan_iax2 scheduler handling. Related to + rev 171452 which resolved the same issues in 1.4. Code from + team/russell/sched_thread2 Review: + http://reviewboard.digium.com/r/129/ + + * main/manager.c: Resolve a memory leak that would occur on an + invalid channel given to Action: Status + +2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com> + + * configs/extensions.conf.sample: 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: 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 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_voicemail.c: 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 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: 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 + ........ + +2009-02-05 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_sqlite.c: Change the first field, or we don't get + the necessary field separation. + +2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_mixmonitor.c: 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 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 ........ + + * apps/app_queue.c: 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 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com> + + * res/res_jabber.c: Add XML documentation for the applications and + functions in res_jabber (closes issue #14405) Reported by: snuffy + Patches: xml_jabber.diff uploaded by snuffy (license 35) + +2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com> + + * channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with + IAX2 transfer not handing off calls. Reverts changes in 116884 + 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. The changes reverted in 116884 caused backwards + compatibility issues involving iax2 transfer with 1.6.0, 1.4, and + 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel + +2009-02-04 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c, include/asterisk/features.h: 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 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com> + + * main/tcptls.c: 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 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_chanspy.c: 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 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: 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 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, pbx/pbx_config.c: 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 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Broke up the large conditional blocks so + it is easy to see if a function is compiled. + +2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/xml.c, include/asterisk/compiler.h, apps/app_stack.c, + include/asterisk/optional_api.h: 1. Make OS X compile cleanly + with app_stack. 2. Use curl to download sound files, as curl is + installed natively on OS X, whereas wget and fetch are not. + (closes issue #14332) Reported by: oej Tested by: Corydon76 + + * /, configs/extensions.conf.sample: 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 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com> + + * /, main/features.c: 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 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com> + + * main/manager.c, CHANGES: Reverting commit number 173028 as there + are some potential issues + + * main/manager.c, CHANGES: Add a CLI command to log out a manager + user (closes issue #13877) Reported by: eliel Patches: + cli_manager_logout.patch.txt uploaded by eliel (license 64) + Tested by: eliel, putnopvut + +2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com> + + * /: Recorded merge of revisions 172962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) + | 11 lines channels/chan_dahdi.c * Added doxygen comments to the + major dahdi structures. * Fixed PRI using an incorrect string + value if the extension delimiter is not present in the Dial() + function. * Fixed some uninitialized string variables on FXS + ports. configs/chan_dahdi.conf.sample * Updated some + documentation. These changes are already in trunk -r172400 + ........ + +2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/features.c, CHANGES, + include/asterisk/features.h: This reverts the changes I made for + 11583; will reviewboard this before committing again... reopened + 11583 until all Russell's issues are resolved. + +2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com> + + * configs/res_ldap.conf.sample: 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 + +2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/features.c, CHANGES, + include/asterisk/features.h: This change allows the disconnect + feature (as in "one-touch" in features.c) to be used within the + dial app, before a call is bridged. Many thanks to sobomax for + submitting this patch. Quoting from bug 11582: "So the goal of + the patch was to use the user configured feature code during the + call setup phase. The original ast_feature_interpret() function + is not well suited for this purpose as it uses much call bridge + specific data and doesn't separate a detection of feature from a + feature handler call. So a new function ast_feature_detect() has + been extracted off the ast_feature_interpret() function but + keeping the original logic intact except some insignificant + changes to locking. "Having created the ast_feature_detect() + function the possibility to use feature detection in almost any + place of the asterisk code. So a call to this function has been + added to wait_for_answer() function of app_dial.so module. This + code doesn't call the feature handler however and uses old call + leg disconnect logic to make the changes as small and simple as + possible to prevent unexpected problems. A disconnect feature + currently is the only one supported during call setup as other + features as call parking and call transfer don't make much sense + during call setup. However if need in some of the features would + arise it is much easier to implement as the infrastructure + changes are already in place with this patch." I have cleaned up + the patch somewhat, and verified that the existing functionality + is not harmed, and that the new functionality works. Terry has + committed his stuff, and there were no conflicts (see 14274). + (closes issue #11583) Reported by: sobomax Patches: + patch-apps__app_dial.c uploaded by sobomax (license 359) + patch-include__asterisk__features.h uploaded by sobomax (license + 359) patch-res__res_features.c uploaded by sobomax (license 359) + enable-features-during-call-setup.diff uploaded by sobomax + (license 359) 11583.newdiff uploaded by murf (license 17) + enable-features-during-call-setup-1.diff uploaded by sobomax + (license 359) 11583.latest-patch uploaded by murf (license 17) + Tested by: sobomax, murf + +2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Fix a spelling mistake. + +2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Add a todo. I do need to really check what's + going on with this kill-the-user business ;-) Why do we suddenly + have two flags to set peer type? + + * channels/chan_sip.c: Small formatting change + + * channels/chan_sip.c: Add some well-needed improvements to the + wishlist in the code, so that we can close some bug reports. + +2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com> + + * channels/chan_sip.c: The CID lookup feature wasn't actually + working properly with dialog-info+xml supporting devices. The + devices (snoms, specifically) need to receive a SIP URI instead + of just an extension. This adds that functionality. + +2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Blank argument crashes Asterisk (closes + issue #14377) Reported by: amorsen + + * funcs/func_strings.c: 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) + +2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/channel.h: Fix redefinition of flag in channel.h + +2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com> + + * configs/features.conf.sample: Remove incorrect line from sample + config + + * apps/app_dial.c, main/global_datastores.c, main/features.c, + include/asterisk/global_datastores.h, CHANGES, + configs/features.conf.sample: 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 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_aes.c: Parameter position reversed in documentation + + * /, autoconf/ast_func_fork.m4, configure, main/app.c, + apps/app_rpt.c, main/asterisk.c: 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 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com> + + * main/cli.c: Remove tabs from comment + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: + 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. + + * include/asterisk/say.h: Fixed some doxygen comments + +2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net> + + * channels/chan_local.c: Revert two lines that was extra, but only + on fridays. + + * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, + include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered + elsewhere" through app_queue with members in chan_local. Also, + implement a private cause code (as suggested by Tilghman). This + works with chan_sip, but doesn't propagate through chan_local. + +2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com> + + * configs/func_odbc.conf.sample: Better document mode=multirow, + based upon a conversation with Jared. + +2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/realtime_pgsql.sql: 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 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample, CHANGES: Update documentation + + * include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make + sure we set setvar= variables on outbound calls too, not only + inbound calls. - Also, change a function in app.c to return a + userful value instead of always returning 0. Patch by fnordian, + changed by Corydon76 and myself. This does not close the bug + report, as fnordian had an additional change we're still + discussing. (related to issue #14059) Reported by: fnordian + Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) + 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) + Tested by: fnordian, Corydon76, oej + + * channels/chan_sip.c: Make sure register= line supports both port + and expiry at the same time. (closes issue #14185) Reported by: + Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by + Nick (license 657) Tested by: Nick_Lewis + + * /, channels/chan_sip.c: 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 22:52 +0000 [r172132] Steve Murphy <murf@digium.com> + + * channels/chan_misdn.c: A further correction: cast the sizeof to + an int. + +2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_odbc.c: Fix how we skip fields (to avoid fields + which don't exist) when doing an UPDATE. (closes issue #14205) + Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt + uploaded by Corydon76 (license 14) Tested by: blitzrage + +2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com> + + * channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 + didn't like the \%ld and issued a warning, breaking my dev-mode + build. This fixes it. + + * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, + main/features.c, include/asterisk/channel.h: 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 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: 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 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net> + + * CHANGES: Yep. Documentation is important. + + * apps/app_queue.c: Add final part of previously committed work for + answered elsewhere in queue - the missing piece that started with + app_dial() earlier on. This is to avoid having the list and + counter of missed calls being touched by queue calls. Add the C + option to queue() and nothing will be logged on phones that + support the Reason: header on SIP cancel, like the SNOM phones. + + * configs/sip.conf.sample: Add some more notes about device + matching. + + * /, configs/sip.conf.sample: 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. + ........ + +2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com> + + * funcs/func_aes.c: Fix some signedness problems in func_aes.c + +2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Don't complain about lack of D-channels on + PTMP connections + +2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com> + + * funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and + AES_DECRYPT dialplan functions. (closes issue #14301) Reported + by: amorsen review: http://reviewboard.digium.com/r/128/ + +2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com> + + * channels/chan_agent.c: 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 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: 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 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there + are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis + Tested by: dbailey + +2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: 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 + ........ + + * channels/chan_sip.c: Moving generic setting to friends + + * channels/chan_sip.c: Continue to move variables into the sip_cfg + structure to make them easier to handle in the future as a group + of settings for a group of devices. At some point, I want one + sip_cfg per domain handled, so we can have "group" settings. + + * channels/chan_sip.c: Just moving around variable declarations so + that we have all globals in the same place. Default setting is + set before we activate the channel or at reloads, not where we + declare the variable. + + * /, channels/chan_sip.c: 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 ........ + + * main/channel.c: Add extensions and context on manager event when + new channel is created. + +2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_oss.c: 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 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: dont segfault when a MWI event occurs on + a line without a registered device + + * configs/skinny.conf.sample: Make the sample skinny.conf work + (closes issue #14325) Reported by: DEA Patches: + skinny.conf.sample-trunk.txt uploaded by DEA (license 3) + +2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com> + + * /, apps/app_page.c: 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 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com> + + * include/asterisk/utils.h: 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. + + * res/res_agi.c: Add a todo to finish the XML docs in this module + +2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com> + + * /, configs/res_odbc.conf.sample: 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 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com> + + * doc/tex/Makefile: Fix asterisk.pdf generation if branch name has + an underscore in it. + +2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com> + + * doc/tex/Makefile: Don't blow up if a branch name has an + underscore in it + +2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com> + + * /, configs/res_odbc.conf.sample: 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 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 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: 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 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 + Jan 2009) | 2 lines Additions to AST-2009-001 ........ + +2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 170568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 + lines When a call is forwarded stop any active indications. The + new channel will provide an indication, if need be, itself. + (closes issue #14310) Reported by: RadicAlish ........ + + * /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 + lines Use the on hold flag to see if the call is on hold or not. + It is possible that our address for them will still be valid even + though they are on hold. (closes issue #14295) Reported by: + klaus3000 ........ + +2009-01-23 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_h323.c: let's use SENTINEL where needed + +2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Reset the ast_str used for escape + substitution. We need to do this since it is a thread local + variable that may contain the value of a previous substitution. + (closes issue #14312) Reported by: pj + +2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: We should not do restart messages if we're + in PTMP mode + +2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: Dont clear the display of skinny phones + when not needed. (closes issue #13182) Reported by: pj Patches: + 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak + (license 7) Tested by: mvanbaak, pj + +2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: MWI messages included in CID spill was not + being properly handled and prevented the call from being + processed (issue #14313) Reported by: seandarcy Tested by: + dbailey + +2009-01-23 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 170392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan + 2009) | 28 lines Fix broken call pickup There was a subtle change + in ast_do_masquerade which resulted in failed attempts to pickup + calls. The problem was that the value of the AST_FLAG_OUTGOING + flag was copied from the clone to the original channel. In the + case of call pickup, this meant that the AST_FLAG_OUTGOING flag + ended up being cleared on the channel that was attempting to + execute the pickup. Because this flag was not set, when ast_read + came across an answer frame, it ignored it. The result of this + was that the calling channel was never properly answered. This + fix changes the behavior in ast_do_masquerade to set the flags on + the original channel to the union of the flags on the clone + channel. This way, if the AST_FLAG_OUTGOING flag is set on either + of the two channels involved in the masquerade, the resulting + channel will have the flag set as well. (closes issue #14206) + Reported by: francesco_r Patches: 14206.patch uploaded by + putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut + ........ + +2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Make sure we don't set the channel to be + inalarm for a D-channel drop on PTMP connections + +2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com> + + * main/abstract_jb.c: Create logfile safely. (closes issue #14160) + Reported by: tzafrir Patches: 20090104__bug14160.diff.txt + uploaded by Corydon76 (license 14) + +2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 170239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 + lines Don't crash if RTCP is not enabled on an RTP structure but + statistics are output. (closes issue #14234) Reported by: jcovert + Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) + rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ + +2009-01-22 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com> + + * /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) + | 6 lines Allow global variables after substitution to be as long + as other variables. (closes issue #14263) Reported by: markd + Patches: 20090120__bug14263.diff.txt uploaded by Corydon76 + (license 14) ........ + +2009-01-22 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 + lines If we are unable to request a DAHDI pseudo channel and we + are using the user introduction without review option make sure + it gets unset so other code does not blindly assume a DAHDI + pseudo channel exists. (closes issue #14282) Reported by: + cheesegrits ........ + +2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: change VMWI to + use new DAHDI_VMWI ioctl call. Change configure script to detect + the new ioctl call data structure. (issue #14104) Reported by: + alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded + by dbailey (license ) Tested by: alecdavis, dbailey + +2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: Merged revisions 170050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 + lines Do a string comparison instead of pointer comparison since + some people specify the context they are actually in as an + argument to get around some funkiness. (closes issue #14011) + Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga + (license 665) ........ + + * apps/app_parkandannounce.c: Clear the autoloop flag when parsing + and setting the context/extension/priority to go back to. When + the channel executes a PBX again we want it to start out at the + point we explicitly say and at that point it will not yet be + doing autoloop. (closes issue #14304) Reported by: jcovert + +2009-01-22 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: * Adjust some conditionals to balance + curly braces. * Other minor changes. + +2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/linkedlists.h: Merged revisions 169943 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) + | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really + wanted to ask is whether autoconf detected a static initializer + value. This fixes rwlocks on all such platforms (mainly, Mac OS + X). (closes issue #13767) Reported by: jcovert Patches: + 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) + Tested by: jcovert, Corydon76 ........ + +2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Whitespace changes only + +2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, /: Merged revisions 169867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 + lines Read lock the contexts to maintain the locking order when + we are notified that the state of a device has changed. (closes + issue #13839) Reported by: mcallist ........ + +2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c: Test commit for test issue #14303 + + * main/say.c: Fix a crash when saying certain numbers in Chinese + This commit fixes a crash that was occurring when attempting to + say a number between 10000 and 100000 due to dividing by 0. This + also removes some places where a "zero" is spoken when it should + not be. (closes issue #14291) Reported by: dant Patches: + say.c-14291.diff uploaded by dant (license 670) Tested by: dant + +2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info> + + * doc/tex/extensions.tex: remove duplicated sentence. + +2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Further fix some oddities in sip show users + and sip show peers logic ccesario on IRC pointed out that his sip + peers were not displayed properly when he would issue the command + "sip show peers." The problem was that the onlymatchonip field + was used to determine if the endpoint was a "peer" or "user." The + tricky part is that a "friend" is supposed to be treated as both + a "user" and a "peer" but the logic would not allow "friends" to + show up as "peers" since onlymatchonip was set to FALSE for + friends. I have modified the sip_peer structure to more + explicitly keep track of what type endpoint it is so that the + various manager and CLI commands will display the expected + information Reported by ccesario via IRC Tested by ccesario + +2009-01-21 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 169722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) + | 8 lines Extra NULLs in the output cause some terminal types to + abort in the middle of a color code, causing terminal weirdness. + (closes issue #14130) Reported by: coolmig Patches: + 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, coolmig ........ + +2009-01-21 17:21 +0000 [r169673] Steve Murphy <murf@digium.com> + + * utils/refcounter.c: This patch corrects a segfault reported in + 14289, due to a null ptr being refd. Yes, seanbright is right in + the bug comments, that is the fix. Sorry for this oversight; I + guess my personal usage didn't have this happen! murf (closes + issue #14289) Reported by: jamesgolovich + +2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant <russell@digium.com> + + * /: Remove properties that erroneously got merged into trunk + + * main/tcptls.c: Fix a regression in TCP support. This patch fixes + a problem that caused chan_sip to think that every open TCP + session was to a remote address of 0.0.0.0:0. (closes issue + #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt + uploaded by jamesgolovich (license 176) + +2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix device state parsing issues for channel + names with multiple slashes The fix being applied is a bit + different for trunk and the 1.6.X branches. For trunk, we only + wish to strip off the characters beyond the second slash if the + channel is a Local channel (i.e. we are removing the /n from the + device name). Other channel technologies with multiple slashes + (e.g. DAHDI) need the information after the second slash in order + to get the proper device state information. In addition to this + fix, the 1.6.X branches are receiving a much more important fix + as well. The problem in 1.6.X is that the member's device name + was being directly changed instead of having a copy changed. This + meant that we would strip off the second slash and trailing + characters and then leave the member's device name like that + permanently thereafter. (closes issue #14014) Reported by: + kebl0155 Patches: 14014_number2.patch uploaded by putnopvut + (license 60) Tested by: kebl0155 + + * apps/app_queue.c: Use the default timeout for a queue instead of + -1 (closes issue #14272) Reported by: timking + + * /, channels/chan_sip.c: Convert the character pointers in a + sip_request to be pointer offsets When an ast_str expands to hold + more data, any pointers that were pointing to the data prior to + the expansion will be pointing at invalid memory. This change + makes such pointers used in chan_sip.c instead be offsets from + the beginning of the string so that the same math may be applied + no matter where in memory the string resides. To help ease this + transition, a macro called REQ_OFFSET_TO_STR has been added to + chan_sip.c so that given a sip_request and an offset, the string + at that offset is returned. (closes issue #14220) Reported by: + riksta Tested by: putnopvut Review + http://reviewboard.digium.com/r/126/ + +2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com> + + * main/features.c: Make a proper builtin attended transfer to + parking work This is an ugly hack from 1.4 that allows the + timeout callback from a parked call to use the right channel name + for the callback when the park is done with a builtin attended + transfer (that isn't completed early). This hasn't ever worked in + trunk and no one has complained yet, so eh. + + * /, main/features.c: Merged revisions 169485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) + | 6 lines Don't play audio to the channel if we've masqueraded + (closes issue #14066) Reported by: bluefox Tested by: + otherwiseguy, bluefox ........ + +2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/res_odbc.h, funcs/func_odbc.c, + include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is + *not* part of the ast_str API, it's part of the ast_odbc API and + just happens to use an ast_str as the buffer; move all of it to + res_odbc.c and res_odbc.h, renaming appropriately along the way + fix some minor coding style issues in strings.h and add some + attribute_pure annotations to functions in the ast_str API + +2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info> + + * main/asterisk.c: fix assignment in swapmode plug. Spotted and fix + provided by ys (closes issue #14129) Reported by: ys Tested by: + ys + + * channels/chan_skinny.c: Redo the event-based MWI in chan_skinny. + Dan saw regular segfaults with the old implementation and rewrote + it to make it really eventbased. I altered it to be trunk + compatible and wedhorn gave some feedback and ideas how to make + it even better. (closes issue #13821) Reported by: DEA Patches: + chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by: + mvanbaak, DEA "no probs by me" from wedhorn + +2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /, apps/app_userevent.c: Merged revisions 169364 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) + | 4 lines Truncate userevents at the end of a line, when the + command exceeds the buffer. (closes issue #14278) Reported by: + fnordian ........ + +2009-01-19 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info> + + * main/asterisk.c: Make asterisk compile on non-amd64 versions of + OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD + and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when + HW_PHYSMEM64 is not available. (closes issue #14129) Reported by: + ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak + (license 7) Tested by: mvanbaak, jtodd + +2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: Get rid of magic number and replace with + DAHDI_VMWI_NUMBER_MASK when determining the number of messages + pending for MWI call + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add + enhanced MWI generation to take advantage of new dahdi line + reversal MWI ability. (closes issue #14104) Reported by: + alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey + (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis + (license 585) Tested by: alecdavis, dbailey + +2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 169210 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, + 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a + potential NULL pointer dereference Move the check for if both + channels on a local_pvt have generators to below where p->chan is + checked for NULLity (NULLness?). This prevents a crash from + occurring if p->chan is NULL. (closes issue #14189) Reported by: + sascha Patches: 14189.patch uploaded by putnopvut (license 60) + Tested by: sascha ........ + +2009-01-17 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add + discriminator for when ring pulse alert signal is used to preface + MWI spills This prevents the situation when MWI messages are + added to caller ID spills causing the channel to be hung up + +2009-01-17 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com> + + * pbx/pbx_dundi.c: Change intializer types. Found while working on + asterisk-cpp. I have a new favorite error message from g++: + pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated + initializers not supported I like it when compilers are + apologetic. + +2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com> + + * main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix + qualify for TCP peer (closes issue #14192) Reported by: + pabelanger Patches: asterisk-bug14192.diff.txt uploaded by + jamesgolovich (license 176) Tested by: jamesgolovich + + * channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when + outboundproxy used (closes issue #14233) Reported by: chris-mac + Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich + (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy + +2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan + 2009) | 18 lines Account for possible NULL pointer when we + receive a 408 in response to a REGISTER It may be that by the + time we receive a reply to a REGISTER request, the attempt has + timed out and thus the registry structure pointed to by the + corresponding sip_pvt has gone away. This situation was handled + properly for a 200 OK response, but the 408 case assumed that the + sip_registry struct was non-NULL, thus potentially causing a + crash This commit fixes this assumption and prints out a message + to the console if we should receive a late 408 response to a + REGISTER (closes issue #14211) Reported by: aborghi Patches: + 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi + ........ + +2009-01-16 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 168716 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) + | 12 lines Convert call to park_call_full to + masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE + return value, we need to use masqueraded parking, otherwise we + will try to call ast_hangup() in __pbx_run() and in + do_parking_thread() and then promptly crash. (closes issue + #14215) Reported by: waverly360 Tested by: otherwiseguy (closes + issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ + +2009-01-16 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com> + + * res/res_timing_timerfd.c: Fix a logic error that occur when using + the timerfd interface This sequence of events posed a problem + timerfd_timer_open timerfd_timer_enable_continuous + timerfd_timer_set_rate timerfd_timer_disable_continuous The + reason was that the timing module was written under the + assumption that timerfd_timer_set_rate would not be called + between enabling and disabling continuous mode. What happened in + this situation was that timerfd_timer_enable_continuous saved off + our previously set timer (in this situation a 0 timer, meaning it + never runs out). Then timerfd_timer_disable_continuous would + restore this 0 timer, even though it logically should set the + timer to be whatever was set in timerfd_timer_set_rate. Now the + behavior in timerfd_timer_set_rate is to overwrite the saved + timer that may or may not have been set in + timerfd_timer_enable_continuous. Even if + timerfd_timer_enable_continuous has not been previously called, + this will not harm the operation. Thanks to Terry Wilson for + discovering the problem and giving me a really great debug + capture that pointed out the problem clearly + +2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c: + Merged revisions 168828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) + | 6 lines Fix the conjugation of Russian and Ukrainian languages. + (related to issue #12475) Reported by: chappell Patches: + vm_multilang.patch uploaded by chappell (license 8) ........ + +2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com> + + * CHANGES: Fix a spelling mistake. + + * channels/chan_misdn.c: build in dev mode + +2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 168745 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | + 14 lines This patch fixes a problem where a goto (or jump, in + this case) fails a consistency check because it can't find a + matching extension. The problem was a missing instruction to end + the range notation in the code where it converts the pattern into + a regex and uses the regex code to determine the match. I tested + using the AEL code the user supplied, and now, the consistency + check passes. (closes issue #14141) Reported by: dimas ........ + + * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch + allows null args in ast_expr2 func calls, and fixes commas being + converted to pipes, which was 1.4 type stuff. If the user says + count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it + won't complain about the empty arg (c,,...) and fabled's patch + won't let it swap the commas for pipes. Ran it thru my dialplan + and no complaints. (closes issue #14169) Reported by: fabled + Patches: function-argument-separator-fix.diff uploaded by fabled + (license 448) + +2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com> + + * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, + funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c, + cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c, + apps/app_voicemail.c: remove the PBX_ODBC logic from the + configure script, and add GENERIC_ODCB logic that includes + copying the relevant LIB and INCLUDE data from either UnixODBC or + iODBC, based on which was found; if both were found, prefer + UnixODBC this stops modules from being linked against both sets + of libraries on systems that have both installed + +2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Add missing brace + + * channels/chan_sip.c: Fix the compactheaders option in sip.conf + + * channels/chan_sip.c: Remove an unneeded condition for line + addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the + sip_request structure had a statically allocated buffer to hold + the text of the request. There was a check in the add_line + function to not attempt to write the line into the buffer if we + did not have room for it. In trunk and Asterisk versions starting + with 1.6.1, an expandable ast_str structure is used to hold the + text. Since it may grow to fit an arbitrarily sized string, this + check in add_line is no longer valid. I found this oddity while + attempting to fix issue #14220; however, I do not believe that + this is the fix for that issue since the output supplied by the + reporter did not contain the warning message that would be + printed had this condition been satisfied. + +2009-01-15 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net> + + * /, configs/extconfig.conf.sample: Merged revisions 168721 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 + lines Meetme actually has realtime but wasn't documented ........ + +2009-01-15 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/strings.h: Resolve issue with negative vs + non-negative length parameters. (closes issue #14245) Reported + by: dveiga + +2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Make sure that we have the same terminology + in sip.conf.sample and the source code warning. Thanks Nick Lewis + for pointing this out in the bug tracker. + + * configs/sip.conf.sample: Clarify some misunderstandings and make + it even more clear that you can refer to a peer in the register= + line. + +2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com> + + * apps/app_meetme.c: Add a missing unlock and properly handle the + 'maxusers' setting on MeetMe conferences. We were using the 'user + number' field to compare against the maximum allowed users, which + works assuming users with lower user numbers didn't leave the + conference. (closes issue #14117) Reported by: sergedevorop + Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright + (license 71) Tested by: sergedevorop + +2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net> + + * CREDITS, CHANGES: Related to issue #14246 Update changes for + SIPRemoveHeader() + + * channels/chan_sip.c: Add capability to remove added SIP headers + *before* INVITE is generated. (closes issue #14246) Reported by: + klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt + uploaded by klaus3000 (license 65) + + * apps/app_queue.c: Add support for setting the Reason header when + cancelling a call in the queue because someone else answered. + Previously, only dial() was supported. EDV-102 + +2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 168628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan + 2009) | 16 lines Fix some crashes from bad datastore handling in + app_queue.c * The queue_transfer_fixup function was searching for + and removing the datastore from the incorrect channel, so this + was fixed. * Most datastore operations regarding the + queue_transfer datastore were being done without the channel + locked, so proper channel locking was added, too. (closes issue + #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by + putnopvut (license 60) Tested by: ZX81, festr ........ + +2009-01-14 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com> + + * main/cli.c: Don't crash when typing 'core set verbose' or 'core + set debug' by themselves. (closes issue #14219) Reported by: + jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded + by jamesgolovich (license 176) + +2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com> + + * /, channels/misdn/isdn_lib.c: Merged revisions 168622 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) + | 4 lines * Fixed create_process() allocation of process ID + values. The allocated process IDs could overflow their respective + NT and TE fields. Affects outgoing calls. ........ + +2009-01-14 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: This fixes a problem where MWI FSK spills + were being injected onto off hook fxs lines. (closes issue + #14143) Reported by: alecdavis Patches: + chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested + by: alecdavis + +2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com> + + * /, contrib/scripts/autosupport: Merged revisions 168614 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan + 2009) | 9 lines Update autosupport script to supply info for both + Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x + and trunk instead of zttest. (closes issue #14132) Reported by: + dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by + dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded + by dsedivec (license 638) ........ + +2009-01-14 20:51 +0000 [r168613] Steve Murphy <murf@digium.com> + + * /, apps/app_page.c: Merged revisions 168608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 + line app_page was failing to compile in dev-mode on my gcc-4.2.4 + system. This change gets rid of the warning. ........ + +2009-01-14 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Restore the "sip show users" and "sip show + user" CLI commands (closes issue #14180) Reported by: amorsen + Patches: sip_show_users_161v3.diff uploaded by putnopvut (license + 60) Tested by: blitzrage, amorsen + +2009-01-14 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info> + + * main/asterisk.c: Fix compilation on FreeBSD and OSX This started + as work to fix the 'core show sysinfo' CLI command but while + working on it oej pointed out that read_credentials did not + compile neither. So while being there, fix that as well. Thanks + for all the testing oej! (closes issue #14129) Reported by: ys + Tested by: oej, mvanbaak + +2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 168603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) + | 7 lines Don't read into a buffer without first checking if a + value is beyond the end. (closes issue #13600) Reported by: atis + Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 + (license 14) Tested by: atis ........ + + * channels/chan_misdn.c: Mostly spacing changes; no functionality + change at all. + +2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com> + + * /, apps/app_page.c: Merged revisions 168593 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) + | 20 lines Don't overflow when paging more than 128 extensions + The number of available slots for calls in app_page was hardcoded + to 128. Proper bounds checking was not in place to enforce this + limit, so if more than 128 extensions were passed to the Page() + app, Asterisk would crash. This patch instead dynamically + allocates memory for the ast_dial structures and removes the + (non-functional) arbitrary limit. This issue would have special + importance to anyone who is dynamically creating the argument + passed to the Page application and allowing more than 128 + extensions to be added by an outside user via some external + interface. The patch posted by a_villacis was slightly modified + for some coding guidelines and other cleanups. Thanks, + a_villacis! (closes issue #14217) Reported by: a_villacis + Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch + uploaded by a (license 660) Tested by: otherwiseguy ........ + +2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_misdn.c: Janitor patch for chan_misdn (make channel + variable access safe) (closes issue #12887) Reported by: pputman + Patches: chan_misdn_threadsafe.patch uploaded by pputman (license + 81) + +2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com> + + * res/res_http_post.c: Fully overwrite a same-named file when + uploading (closes issue #14190) Reported by: timking + + * Makefile, include/asterisk/options.h, main/asterisk.c: Add option + to hide console connect messages (closes issue #14222) Reported + by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded + by jamesgolovich (license 176) Tested by: otherwiseguy + +2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Clarify a message that app_queue prints and + change to a debug-level message The "No one is answering..." + verbose message contained 3 numbers that were not explained in + any way to whoever was viewing the message. It is more helpful + now since the message explains what the numbers mean. Also, the + message has been downgraded to "DEBUG" level. (closes issue + #14172) Reported by: caio1982 Patches: queue_answering_debug.diff + uploaded by caio1982 (license 22) + +2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) + | 7 lines Don't pass a value with a side effect to a macro + (closes issue #14176) Reported by: paraeco Patches: + chan_sip.c.diff uploaded by paraeco (license 658) ........ + +2009-01-13 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow + specifying a port number in the user portion of a register => + line in sip.conf With this commit, a register => line in sip.conf + may contain a port number in the "user" section of the line. + Please see CHANGES and sip.conf.sample for more details regarding + this. (closes issue #14198) Reported by: Nick_Lewis Patches: + chan_sip.c-domainport2.patch uploaded by Nick (license 657) + Tested by: Nick_Lewis + +2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com> + + * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /, + include/asterisk/indications.h, apps/app_readexten.c, + apps/app_disa.c, include/asterisk/channel.h, main/indications.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, + funcs/func_channel.c, main/app.c, res/snmp/agent.c, + res/res_indications.c: Merged revisions 168561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) + | 2 lines Revert unnecessary indications API change from rev + 122314 ........ + +2009-01-13 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) + | 6 lines If either conditional is NULL, don't try copying it. + (closes issue #14226) Reported by: caspy Patches: + 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) + ........ + +2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * main/taskprocessor.c: correct a CLI description + +2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 + Dec 2008) | 5 lines Repeat attempts to write when we receive + -EAGAIN from the driver, as detailed in the ALSA sample code (see + http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) + Reported by: Jerry Geis (via the -users list) Fixed by: me + (license 14) ........ + +2009-01-12 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com> + + * main/srv.c: bump the verbosity of a message in srv.c up by one. + It used to be at this level prior to a large patch merge which + converted ast_verbose calls to ast_verb (closes issue #14221) + Reported by: jcovert Patches: srv.c.patch uploaded by jcovert + (license 551) + +2009-01-12 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/app.c: Some platforms (notably, the BSDs) have a more + efficient implementation called closefrom(3). + +2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_agi.c: Merged revisions 168516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) + | 5 lines (closes issue #13881) Reported by: hoowa Update the app + CDR field for AGI commands that are not executing an application + via "exec". ........ + + * /, channels/chan_agent.c: Merged revisions 168507 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 + Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG + Tested by: denisgalvao This gits rid of the notion of an + owning_app allowing the request and hangup to be initiated by + different threads. Originating from an active agent channel + requires this. The implementation primarily changes __login_exec + to wait on a condition variable rather than a lock. Review: + http://reviewboard.digium.com/r/35/ ........ + +2009-01-12 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net> + + * apps/app_minivm.c: Better to use the proper app name + +2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Merged revisions 168482 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan + 2009) | 5 lines I am reverting the fix made in revision 168128 + (and its upward merges) after being contacted by Olle Johansson + and being shown how this fix is incorrect. Thanks to Olle for + clearing this up for me. ........ + +2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com> + + * /, configs/indications.conf.sample: Merged revisions 168480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) + | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ + +2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net> + + * main/asterisk.c: Don't include swap.h unless we have swapctl + +2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for + reconstructing a field value. + +2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com> + + * /, sounds/Makefile: Merged revisions 168267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan + 2009) | 1 line update to use new sound file packages that include + license files ........ + +2009-01-09 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c: Spacing change + +2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info> + + * contrib/scripts/sip_nat_settings (added), CHANGES: Add a script + to find out the correct settings for Asterisk behind NAT (closes + issue #13065) Reported by: tzafrir Patches: sip_nat_settings + uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by + mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and + moi + +2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 + Jan 2009) | 2 lines Make this compile for mvanbaak ........ + +2009-01-09 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan + 2009) | 13 lines Add check_via calls to more request handlers + INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not + checking the topmost Via to determine where to send the response. + Adding check_via calls to those request handlers solves this. + (closes issue #13071) Reported by: baron Patches: check_via.patch + uploaded by baron (license 531) Tested by: baron ........ + +2009-01-09 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 + Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 * + Miscellaneous doxygen comments added. ........ + +2009-01-09 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not + exist (closes issue #14203) Reported by: jamesgolovich Patches: + asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich + (license 176) + +2009-01-09 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c, include/asterisk/strings.h: When using ast_str + with a non-ast_str-enabled API, we need to update the buffer or + otherwise, we cannot use ast_str_strlen(). + +2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com> + + * main/logger.c: Added a comment to logger.c about where to put + includes + + * main/logger.c: Use ast_safe_system() in logger.c instead of + system() (closes issue #14194) Reported by: pabelanger + +2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com> + + * apps/app_originate.c: Set ORIGINATE_STATUS instead of + OUTGOING_STATUS to match the documentation + + * apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN + and MACRO_CONTEXT will be set + +2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 167840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) + | 6 lines Don't truncate database results at 255 chars. (closes + issue #14069) Reported by: evandro Patches: + 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) + ........ + +2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Revert chan_sip changes which were + accidentally committed in revision 167792 + +2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com> + + * apps/app_minivm.c: Fix variables to comply with documentation + changes + + * apps/app_minivm.c: Textual changes, consistency in status + variable naming, and other minor bugs. (closes issue #13943) + Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded + by Marquis (license 32) + +2009-01-08 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average + talk time for a queue This patch adds the functionality to + app_queue of calculating the average amount of time that channels + are bridged for a queue. The algorithm used to calculate the + average is the same exponential average currently used to + calculate the average holdtime. See the CHANGES file to see the + methods you may use to view this information. (closes issue + #13960) Reported by: coolmig Patches: + app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) + +2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, CHANGES: Convert dialplan application + DAHDISendCallreroutingFacility to use commas. (closes issue + #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded + by eliel (license 64) + +2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan + 2009) | 1 line remove an unnecessary argument to queue_request() + ........ + + * channels/chan_sip.c: Merged revisions 167620 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan + 2009) | 5 lines When a SIP request or response arrives for a + dialog with an associated Asterisk channel, and the lock on that + channel cannot be obtained because it is held by another thread, + instead of dropping the request/response, queue it for later + processing when the channel lock becomes available. + http://reviewboard.digium.com/r/123/ ........ + +2009-01-08 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/sip-friends.sql: Oops... fix the fieldname I + changed yesterday to be right. + +2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com> + + * /, main/file.c: Merged revisions 167566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) + | 2 lines Fix the last couple of places where free() was + improperly used directly. ........ + + * /, main/file.c: Merged revisions 167554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) + | 2 lines Don't fclose() the file early, the filestream + destructor will handle it. ........ + + * /, main/file.c: Merged revisions 167545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) + | 2 lines Only try to close the file if one was actually opened + ........ + + * /, main/file.c: Merged revisions 167541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) + | 4 lines Don't use free() directly. This caused a crash since + ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, + #asterisk-dev ........ + +2009-01-07 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com> + + * apps/app_followme.c: Answer the channel if it has not already + been answered and we've already found a valid profile for + followme. (closes issue #14140) Reported by: dimas Patches: + 14140.patch uploaded by dimas + +2009-01-07 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com> + + * configs/queues.conf.sample: Update queues.conf.sample + documentation. Update the queues.conf.sample documentation to + mention that you need to preload chan_local.so as well if you + plan on using Local channels for queue members, and you're + preloading pbx_config.so. (closes issue #14179) Reported by: + CrashHD Tested by: CrashHD + +2009-01-07 17:35 +0000 [r167442] Russell Bryant <russell@digium.com> + + * /, main/indications.c: Merged revisions 167432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) + | 4 lines Treat an empty string the same way as a NULL country + argument. In passing, simplify the handling of returning a + default tone zone. ........ + +2009-01-07 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com> + + * channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected + during mwi spill. Correct logic error in handling events when + sending mwi spill (closes issue #14143) Reported by: alecdavis + Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by + dbailey + +2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/sip-friends.sql: Update the sip-friends.sql file + to use the non-deprecated 'defaultname' instead of 'username' and + remove an extra comma that would cause the script to fail as-is + +2009-01-06 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com> + + * /, main/db.c: Merged revisions 167299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan + 2009) | 8 lines Use the correct variable when creating the format + string (closes issue #14177) Reported by: nic_bellamy Patches: + asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic + (license 299) ........ + +2009-01-06 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600 + (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 + Jan 2009) | 2 lines Security fix AST-2009-001. ........ + ................ + +2009-01-05 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan + 2009) | 41 lines A couple of changes to T.38 SDP attribute + handling There are some boolean attributes for T.38 such as + T38FaxFillBitRemoval, T38FaxTranscodingMMR, and + T38FaxTranscodingJBIG. By simply being present, we should treat + these as a "true" value. The current code, however, was requiring + a 1 or 0 as the value of the attribute in order to parse it. This + is due to the fact that there are some T.38 endpoints and + gateways that also transmit this information incorrectly. This + patch follows the "be liberal in what you accept and strict in + what you send" philosophy by accepting both the correctly- and + incorrectly-formatted attributes, but only sending information as + it is supposed to be sent. It was also discovered that a + particular type of T.38 gateway sends some non-standard T.38 SDP + attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, + it used T38MaxDatagram and T38FaxMaxRate respectively. We now + will properly accept these attributes as well. Note that there + are a lot of patches cited in the below commit message template. + This is because the person who submitted these patches is an + awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes + issue #13976) Reported by: linulin Patches: + chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) + chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov + (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded + by arcivanov (license 648) + chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov + (license 648) Tested by: arcivanov ........ + +2009-01-05 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com> + + * UPGRADE-1.6.txt: More clearly explain that quote marks are no + longer necessary. (closes issue #13718) Reported by: davidw + Patches: 20081020__bug13718.diff.txt uploaded by Corydon76 + (license 14) Tested by: blitzrage + +2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com> + + * main/asterisk.c: When parsing environment variable + ASTERISK_PROMPT, make sure to proceed to the next character when + a non format specifier is used (no %). Otherwise, the while loop + looking for the null byte will never exit. + +2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com> + + * doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README: + Mostly just whitespace, but also convert 'CVS' to 'SVN' in a + couple places and fix a few typos I found in the + CODING_GUIDELINES. + +2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com> + + * main/xmldoc.c: Don't forget to free typename + +2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c: Change some incorrect syntax for pri set + debug and correct an off-by-one error in ss7 set debug command + +2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com> + + * main/ast_expr2.h, main/ast_expr2.c: That was weird... + + * channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c: + Merged revisions 166953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) + | 5 lines Also inherit the musiconhold class. (Closes #14153) + Reported by: Jerry Geis, via the users list. Patch by: me + (license 14) ........ + +2008-12-30 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c, doc/sip-retransmit.txt, + doc/tex/phoneprov.tex, res/res_http_post.c, + phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some + svn:keywords + +2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with + the removal of AST_PBX_KEEPALIVE When placing a call to a queue + which ran a gosub on the member's channel, Asterisk would crash + every time, stemming from the fact that the member's channel was + being hung up unexpectedly when the Gosub completed. The + necessary change was pretty much copied and pasted from + app_dial's similar changes made last week. I also took the + opportunity to change a LOG_DEBUG message in app_dial to use + ast_debug. I am guessing this was due to a direct merge from 1.4 + that was not corrected to use trunk's preferred syntax. + +2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com> + + * funcs/func_audiohookinherit.c: Fix a typo in the XML + documentation of the AUDIOHOOK_INHERIT dialplan function. + +2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com> + + * /, channels/misdn_config.c: Merged revisions 166772 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 + Dec 2008) | 4 lines Use strncat() instead of an sprintf() in + which source and target buffers overlap + http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html + ........ + +2008-12-24 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com> + + * channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I + believe 26.2.2 is what was meant: Note that in the SIPS URI + scheme, transport is independent of TLS, and thus + "sips:alice@atlanta.com;transport=tcp" and + "sips:alice@atlanta.com;transport=sctp" are both valid (although + note that UDP is not a valid transport for SIPS). The use of + "transport=tls" has consequently been deprecated, partly because + it was specific to a single hop of the request. This is a change + since RFC 2543. + +2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Allow semicolons and extended characters in + user-specified SIP headers. (closes issue #14110) Reported by: + gork Patches: 20081222__bug14110__2.diff.txt uploaded by + Corydon76 (license 14) Tested by: gork, putnopvut + +2008-12-23 18:13 +0000 [r166665] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/pbx.c, /, main/features.c, + apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c, + include/asterisk/features.h: Merged revisions 166093 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 In + order to merge this 1.4 patch into trunk, I had to resolve some + conflicts and wait for Russell to make some changes to res_agi. I + re-ran all the tests; 39 calls in all, and made fairly careful + notes and comparisons: I don't want this to blow up some aspect + of asterisk; I completely removed the KEEPALIVE from the pbx.h + decls. The first 3 scenarios involving feature park; feature xfer + to 700; hookflash park to Park() app call all behave the same, + don't appear to leave hung channels, and no crashes. ........ + r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | + 131 lines This merges the masqpark branch into 1.4 These changes + eliminate the need for (and use of) the KEEPALIVE return code in + res_features.c; There are other places that use this result code + for similar purposes at a higher level, these appear to be left + alone in 1.4, but attacked in trunk. The reason these changes are + being made in 1.4, is that parking ends a channel's life, in some + situations, and the code in the bridge (and some other places), + was not checking the result code properly, and dereferencing the + channel pointer, which could lead to memory corruption and + crashes. Calling the masq_park function eliminates this danger in + higher levels. A series of previous commits have replaced some + parking calls with masq_park, but this patch puts them ALL to + rest, (except one, purposely left alone because a masquerade is + done anyway), and gets rid of the code that tests the KEEPALIVE + result, and the NOHANGUP_PEER result codes. While bug 13820 + inspired this work, this patch does not solve all the problems + mentioned there. I have tested this patch (again) to make sure I + have not introduced regressions. Crashes that occurred when a + parked party hung up while the parking party was listening to the + numbers of the parking stall being assigned, is eliminated. These + are the cases where parking code may be activated: 1. Feature one + touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. + Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi + hookflash xfer to 700) 4. Run Park via manager. The interesting + testing cases for parking are: I. A calls B, A parks B a. B hangs + up while A is getting the numbers announced. b. B hangs up after + A gets the announcement, but before the parking time expires c. B + waits, time expires, A is redialed, A answers, B and A are + connected, after which, B hangs up. d. C picks up B while still + in parking lot. II. A calls B, B parks A a. A hangs up while B is + getting the numbers announced. b. A hangs up after B gets the + announcement, but before the parking time expires c. A waits, + time expires, B is redialed, B answers, A and B are connected, + after which, A hangs up. d. C picks up A while still in parking + lot. Testing this throroughly involves acting all the + permutations of I and II, in situations 1,2,3, and 4. Since I + added a few more changes (ALL references to KEEPALIVE in the + bridge code eliimated (I missed one earlier), I retested most of + the above cases, and no crashes. H-extension weirdness. Current + h-extension execution is not completely correct for several of + the cases. For the case where A calls B, and A parks B, the 'h' + exten is run on A's channel as soon as the park is accomplished. + This is expected behavior. But when A calls B, and B parks A, + this will be current behavior: After B parks A, B is hung up by + the system, and the 'h' (hangup) exten gets run, but the channel + mentioned will be a derivative of A's... Thus, if A is DAHDI/1, + and B is DAHDI/2, the h-extension will be run on channel + Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be + those relating to Channel A. And, in the case where A is + reconnected to B after the park time expires, when both parties + hang up after the joyful reunion, no h-exten will be run at all. + In the case where C picks up A from the parking lot, when either + A or C hang up, the h-exten will be run for the C channel. CDR's + are a separate issue, and not addressed here. As to WHY this + strange behavior occurs, the answer lies in the procedure + followed to accomplish handing over the channel to the parking + manager thread. This procedure is called masquerading. In the + process, a duplicate copy of the channel is created, and most of + the active data is given to the new copy. The original channel + gets its name changed to XXX<ZOMBIE> and keeps the PBX + information for the sake of the original thread (preserving its + role as a call originator, if it had this role to begin with), + while the new channel is without this info and becomes a call + target (a "peer"). In this case, the parking lot manager thread + is handed the new (masqueraded) channel. It will not run an + h-exten on the channel if it hangs up while in the parking lot. + The h exten will be run on the original channel instead, in the + original thread, after the bridge completes. See bug 13820 for + our intentions as to how to clean up the h exten behavior. + Review: http://reviewboard.digium.com/r/29/ ........ + +2008-12-23 16:04 +0000 [r166625] Russell Bryant <russell@digium.com> + + * CHANGES: Fix spelling error. + +2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 166568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec + 2008) | 12 lines Fix a crash resulting from a datastore with + inheritance but no duplicate callback The fix for this is to + simply set the newly created datastore's data pointer to NULL if + it is inherited but has no duplicate callback. (closes issue + #14113) Reported by: francesco_r Patches: 14113.patch uploaded by + putnopvut (license 60) Tested by: francesco_r ........ + +2008-12-23 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 166509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) + | 4 lines Use the integer form of condition for integer + comparisons. (closes issue #14127) Reported by: andrew ........ + +2008-12-22 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com> + + * res/res_agi.c: Always use the value of the AGISIGHUP when running + an AGI. Prior to this patch, the value of AGISIGUP was not always + honored when set on a channel. (closes issue #13711) Reported by: + fmueller Patches: 13711.patch uploaded by putnopvut (license 60) + +2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com> + + * res/res_musiconhold.c: Cosmetic change - don't mix struct + initializer styles. + +2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, + 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks + and autoservice It has been discovered that if a channel is + locked prior to a call to ast_autoservice_stop, then it is likely + that a deadlock will occur. The reason is that the call to + ast_autoservice_stop has a check built into it to be sure that + the thread running autoservice is not currently trying to + manipulate the channel we are about to pull out of autoservice. + The autoservice thread, however, cannot advance beyond where it + currently is, though, because it is trying to acquire the lock of + the channel for which autoservice is attempting to be stopped. + The gist of all this is that a channel MUST NOT be locked when + attempting to stop autoservice on the channel. In this particular + case, the channel was locked by a call to ast_read. A call to + ast_exists_extension led to autoservice being started and stopped + due to the existence of dialplan switches. It may be that there + are future commits which handle the same symptoms but in a + different location, but based on my looks through the code, it is + very rare to see a construct such as this one. (closes issue + #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded + by putnopvut (license 60) Tested by: rtrauntvein Review: + http://reviewboard.digium.com/r/107/ ........ + +2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com> + + * res/res_musiconhold.c: Fix a bad typo. + + * main/astobj2.c: Remove some error messages. This is the default + handler that is valid to use. + + * /, main/utils.c: Merged revisions 166297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) + | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ + + * include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce + ast_careful_fwrite() and use in AMI to prevent partial writes. + This patch introduces a function to do careful writes on a file + stream which will handle timeouts and partial writes. It is + currently used in AMI to address the issue that has been + reported. However, there are probably a few other places where + this could be used. (closes issue #13546) Reported by: srt Tested + by: russell http://reviewboard.digium.com/r/104/ + + * res/res_musiconhold.c: Re-work ref count handling of MoH classes + using astobj2 to resolve crashes. (closes issue #13566) Reported + by: igorcarneiro Tested by: russell Review: + http://reviewboard.digium.com/r/106/ + +2008-12-22 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com> + + * main/dnsmgr.c: Record the previous port in the temporary address + structure so that the comparison does not treat the host as + having changed even if it did not. This would have been + uninitialized before and would have led to a baddddd port. + (closes issue #13628) Reported by: pananix Patches: + bug13628.patch uploaded by jpeeler (license 325) Tested by: file, + blitzrage + +2008-12-22 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com> + + * funcs/func_timeout.c, main/file.c: Fix a file playback crash and + explicitly initialize values in func_timeout.c A crash was + brought up on the bugtracker. The first run through valgrind was + full of legitimate complaints of uninitialized values in + func_timeout when setting a response timeout. These were fixed + but the crash persisted. A second run through showed the real + problem. The reference counting used for filestreams was + incorrect because there were some missing increments when a frame + was read from a format module. (closes issue #14118) Reported by: + blitzrage Patches: 14118v2.patch uploaded by putnopvut (license + 60) Tested by: blitzrage + +2008-12-22 14:16 +0000 [r166258] Russell Bryant <russell@digium.com> + + * res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This + patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The + only usage was for the AGI command, "asyncagi break". This patch + removes this feature. Normally, a feature would not be removed + like this. However, this code is broken and usage of it will + result in a memory leak. Usage of this feature will make the AGI + code return a result of AST_PBX_KEEPALIVE. The PBX handler + assumes that another thread has assumed ownership of the channel. + The channel thread will exit without destroying the channel. + Unfortunately, _no_ thread has ownership of the channel at this + point. There are a couple of serious problems here: 1) The only + way to recover the caller is to issue a channel redirect. This + will work, but this will be done with a masquerade, and the old + ast_channel structure will be lost. 2) Until the channel redirect + happens, there is no code servicing the channel. That means + nothing is reading audio or handling events coming from the + channel. This is very bad. The recommended way to get this same + "break" functionality is to issue the redirect while the channel + is still being handled by the AGI code. That way, there will be + no memory leak, and there will be no period of time that the + channel is not being serviced. + +2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com> + + * include/asterisk/doxyref.h: Make a note about formatting the + review URL in commit messages + +2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com> + + * main/audiohook.c: Get rid of an extra space. I don't know how + this crept back in when I had already fixed it earlier + + * funcs/func_audiohookinherit.c: Remove the verbatim tag from the + author line I could have sworn I already did that before, + though... + + * main/channel.c, funcs/func_audiohookinherit.c (added), + include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a + new dialplan function AUDIOHOOK_INHERIT This function is being + added as a method to allow for an audiohook to move to a new + channel during a channel masquerade. The most obvious use for + such a facility is for MixMonitor when a transfer is performed. + Prior to the addition of this functionality, if a channel running + MixMonitor was transferred by another party, then the recording + would stop once the transfer had completed. By using + AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the + call even after the transfer has completed. It has also been + determined that since this is seen by most as a bug fix and is + not an invasive change, this functionality will also be + backported to 1.4 and merged into the 1.6.0 branches, even though + they are feature-frozen. (closes issue #13538) Reported by: mbit + Patches: 13538.patch uploaded by putnopvut (license 60) Tested + by: putnopvut Review: http://reviewboard.digium.com/r/102/ + +2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Add configuration + support for half_full DAHDI buffer policy + +2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com> + + * apps/app_record.c: Fix the XML documentation for Record(). + <value> tags inside <variable> elements must have CDATA and no + another XML node. + +2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) + | 9 lines Ensure that the chanspy datastore is fully initialized. + This patch resolved some random crash issues observed by a user + on a BSD system (closes issue #14111) Reported by: ys Patches: + app_chanspy.c.diff uploaded by ys (license 281) ........ + + * include/asterisk/doxyref.h: Disable some automatic links + generated by doxygen. + + * include/asterisk/doxyref.h: Introduce commit message formatting + guidelines. This documents the recommended outline to use for + commit message. It also covers information on special tags that + can be used in commit messages, as well as the template + functionality that is available on bugs.digium.com. Review: + http://reviewboard.digium.com/r/96/ + + * /, main/utils.c: Merged revisions 165796 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) + | 11 lines Make ast_carefulwrite() be more careful. This patch + handles some additional cases that could result in partial writes + to the file description. This was done to address complaints + about partial writes on AMI. (issue #13546) (more changes needed + to address potential problems in 1.6) Reported by: srt Tested by: + russell Review: http://reviewboard.digium.com/r/99/ ........ + +2008-12-18 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c: (closes issue #13993) Reported by: mika Add + ActionID response to ping if sent with request. + +2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 + Dec 2008) | 8 lines Add mutexes around accesses to the IMAP + library interface. This prevents certain crashes, especially when + shared mailboxes are used. (closes issue #13653) Reported by: + howardwilkinson Patches: + asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by + howardwilkinson (license 590) Tested by: jpeeler ........ + +2008-12-18 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com> + + * channels/chan_dahdi.c, channels/chan_misdn.c, + channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c, + channels/chan_oss.c: Numerous documentation updates. (closes + issue #13970) Reported by: pkempgen Patches: + __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage + (license 10) + +2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com> + + * res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was + being set NULL just prior to being dereferenced in an ao2_link + call. I have moved the setting of the variable to NULL until + after the ao2_link call. + +2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com> + + * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the + need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This + is part of an effort to completely remove AST_PBX_KEEPALIVE and + other similar return codes from the source. While this usage was + perfectly safe, there are others that are problematic. Since we + know ahead of time that we do not want to PBX to destroy the + channel, the PBX API has been changed so that information can be + provided as an argument, instead, thus removing the need for the + KEEPALIVE return value. Further changes to get rid of KEEPALIVE + and related code is being done by murf. There is a patch up for + that on review 29. Review: http://reviewboard.digium.com/r/98/ + + * /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 + Dec 2008) | 7 lines Set the process group ID on the MOH process + so that all children will get killed (closes issue #14099) + Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2 + uploaded by caspy (license 645) ........ + +2008-12-18 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Fix 2 resource leaks and fix another + pipe-to-comma conversion + +2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 165591 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 + lines Only care about a compatible codec for early bridging if we + are actually bridging to another channel. If we are not we + actually want to bring the audio back to us. (closes issue + #13545) Reported by: davidw ........ + +2008-12-18 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: Fix reference counts of the class and add an + assertion to the end. + +2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com> + + * main/strings.c, include/asterisk/strings.h: Remove duplicate code + from the ast_str API. We now use __AST_STR_* to access 'struct + ast_str' members, but this must only be used inside the API + implementation. (closes issue #14098) Reported by: eliel Patches: + ast_str.patch uploaded by eliel (license 64) + +2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com> + + * apps/app_originate.c: Add a \todo note for app_originate. Jared + Smith suggested that we add a way to be able to set variables and + functions on the outbound channel. I think that it's a great + idea, so I have added it as a todo so that it gets done at some + point. + + * apps/app_originate.c (added), CHANGES: Add a new application, + Originate. (closes issue #14075) Reported by: rcasas Patches: + app_originate.c uploaded by rcasas (license 641), heavily + modified by me Tested by: russell Review: + http://reviewboard.digium.com/r/95/ + +2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com> + + * apps/app_record.c: Add RECORD_STATUS variable, as requested on + the -users list. Patch by me (license 14) + +2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson <mmichelson@digium.com> + + * res/res_odbc.c: Fix a refcount leak in res_odbc + + * apps/app_meetme.c, res/res_realtime.c: Fix the build + +2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher <tlesher@digium.com> + + * apps/app_macro.c: Oops, broke trunk + + * /, apps/app_macro.c: Merged revisions 165317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) + | 4 lines Reverse the fix from issue #6176 and add proper + handling for that issue. (Closes issue #13962, closes issue + #13363) Fixed by myself (license 14) ........ + +2008-12-17 21:17 +0000 [r165318] Mark Michelson <mmichelson@digium.com> + + * apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c, + apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec + 2008) | 7 lines Fix some memory leaks found while looking at how + realtime configs are handled. Also cleaned up some coding + guidelines violations in app_realtime.c, mostly related to + spacing ........ + +2008-12-17 20:50 +0000 [r165254] Steve Murphy <murf@digium.com> + + * utils/extconf.c: This patch is here committed to satisfy the + buildbot, who has a problem with the const. + +2008-12-17 19:55 +0000 [r165219] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Polycom phones close the connection after + reading a little bit of the firmware files, we should stop + sending in that case. Also, make that case print out a debug + statement instead of a scary WARNING. + +2008-12-17 19:52 +0000 [r165216] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Call proxy_update so that the IP address + gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue + #14055) Reported by: chris-mac + +2008-12-17 18:49 +0000 [r165180] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 + +2008-12-17 17:56 +0000 [r165145] Russell Bryant <russell@digium.com> + + * doc/appdocsxml.dtd: argsep is used as an attribute for an + argument, as well + +2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: And actually assign the function to a + pointer... + + * apps/app_voicemail.c: Use the create_vm_state_from_user function + in a place where it was not being used before. Also, I've moved + the urgent folder check in messagecount() up a bit so that the + flow is a bit better. This was something I noticed while taking a + look at issue #13973, although I don't think this is the + underlying cause of the issue. + +2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming <kpfleming@digium.com> + + * utils: ignore this copied file + +2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy <murf@digium.com> + + * utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c, + utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix" + for a "horribly broken" situation. As stuff shifts around in the + asterisk code, the miscellaneous inclusions from the standalone + stuff gets broken. There's no easy fix for this situation. I made + sure that everything in utils builds without problem ***AND*** + that aelparse runs the regressions correctly with the following + make menuselect options both on and off: DONT_OPTIMIZE + DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE + DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE + I think from now on, I'm going to #undef all these features in + the various utils native files; I guess I could do the same for + the copied-in files, surrounded by STANDALONE ifdef. A standalone + isn't going to care about threads, mutexes, etc. + + * pbx/ael/ael-test/ref.ael-vtest17, + pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions + +2008-12-16 23:06 +0000 [r164978] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec + 2008) | 7 lines After looking through SIP registration code most + of the day, this is one of the few things I could find that was + just plain wrong. Even though it probably isn't possible for it + to happen, it seems weird to have code that checks if a pointer + is NULL and then immediately dereferences that pointer if it was + NULL. ........ + +2008-12-16 22:57 +0000 [r164976] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, doc/api-1.6.2-changes.txt (added), + funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c, + CHANGES, configs/extensions.conf.sample: Add timezone to the + possible fields in a timespec. (closes issue #14028) Reported by: + mostyn Patches: timezone-v2.patch uploaded by mostyn (license + 398) (with additional code guideline fixes and a memory leak fix + by me - license 14) + +2008-12-16 22:45 +0000 [r164942] Jeff Peeler <jpeeler@digium.com> + + * apps/app_record.c: (closes issue #13669) Reported by: pj Delete + file recording if recording terminated from a hangup. + +2008-12-16 22:31 +0000 [r164941] Terry Wilson <twilson@digium.com> + + * channels/chan_sip.c: Make a note of the feature request in bug + #11157 as per the reporter and oej, and suspend the bug since no + one seems to be keen on implementing it any time soon. + +2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 164881 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) + | 9 lines Fix an issue where DEBUG_THREADS may erroneously report + that a thread is exiting while holding a lock. If the last lock + attempt was a trylock, and it failed, it will still be in the + list of locks so that it can be reported. (closes issue #13219) + Reported by: pj ........ + + * /, apps/app_macro.c: Merged revisions 164876 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) + | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has + been returned. This is a bug I noticed while looking at the code + for app_macro. This return code means that another thread has + assumed ownership of the channel and it can no longer be touched. + (I hate this return code with a passion, by the way.) ........ + + * main/asterisk.c: Fix build issues on Linux after sysinfo related + changes + +2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify + trumps poke per lmadsen. + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + configuration options for finer control over how Asterisk handles + having to poke all peers at seemingly the same time. (closes + issue #13217) Reported by: cervajs + +2008-12-16 20:41 +0000 [r164807] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 164806 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) + | 9 lines Add "restart gracefully" to the AMI blacklist of CLI + commands. "module unload" was already identified as a command + that can not be used from the AMI. "restart gracefully" + effectively unloads all modules, and will run in to the same + problems. (closes issue #13894) Reported by: kernelsensei + ........ + +2008-12-16 20:08 +0000 [r164802] Michiel van Baak <michiel@vanbaak.info> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/asterisk.c: introduce 'core show sysinfo' for systems that + dont have the Linux-ish sysinfo stuff but do have sysctl. (closes + issue #13433) Reported by: mvanbaak Patches: + 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license + 7) with two free calls replaced with ast_free based on feedback + on reviewboard Review: http://reviewboard.digium.com/r/91/ + +2008-12-16 20:04 +0000 [r164801] Steve Murphy <murf@digium.com> + + * main/pbx.c: (closes issue #14076) Reported by: toc Tested by: + murf OK, Well this issue has had its share of flip-flopping. I + found the following: 1. the code in question, in ext_cmp1 in + pbx.c, would not allow two extensions that vary only by any + dashes contained within them, to be defined in the same context. + 2. for input dialstrings, dashes are NOT ignored. So, skipping + them when sorting patterns seemed a bit silly. Thus, you might + declare ext 891 in a context, but if you try dialing 8-9-1, it + will NOT match 891. So, I proposed to remove the code from + ext_cmp1 to skip the spaces and dashes. Just kept us from + declaring 891 and 8-9-1 in the same context, forcing users to + generate otherwise uselessly obfuscated dialplan code to get the + same effect. Then, I tried out 1.4, and found that: 1. you can + declare 891 and 8-9-1 in the same context! 2. You can't define + 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have + 891 match it! So, it appears that my proposal simply restores the + pbx to behaving as it did in 1.4. + +2008-12-16 19:54 +0000 [r164798] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/safe_asterisk: Set up umask as a possible + configuration option. (closes issue #13753) Reported by: irroot + +2008-12-16 17:14 +0000 [r164737] Russell Bryant <russell@digium.com> + + * /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged + revisions 164736 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) + | 14 lines Fix memory leak and invalid reporting issues with + DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was + being used within the context of the thread local data + destructors. We would go off and allocate more thread local data + while the pthread lib was in the middle of destroying it all. + This led to a memory leak. Another issue was an invalid argument + being provided to the the object_add API call. (closes issue + #13678) Reported by: ys Tested by: Russell ........ + +2008-12-16 16:50 +0000 [r164733] Joshua Colp <jcolp@digium.com> + + * pbx/pbx_config.c: Be more detailed about why the include did not + get included. (closes issue #14071) Reported by: kshumard + Patches: pbx_config.patch.improvederroroutput.txt uploaded by + kshumard (license 92) + +2008-12-16 16:00 +0000 [r164675] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) + | 11 lines Fix a memory leak related to the use of the "setvar" + configuration option. The problem was that these variables were + being appended to the list of vars on the sip_pvt every time a + re-registration or re-subscription came in. Since it's just a + waste of memory to put them there unless the request was an + INVITE, then the fix is to check the request type before copying + the vars. (closes issue #14037) Reported by: marvinek Tested by: + russell ........ + +2008-12-16 15:44 +0000 [r164659] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: When using externhost make sure the port + gets set to the bindaddr port if one was not specified in the + externhost value itself. (closes issue #13634) Reported by: + performer + +2008-12-16 15:31 +0000 [r164648] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 164634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 + lines I added a sentence to clarify why - and ' ' are ignored in + patterns as per bug 14076. Leif says he'll put some stuff about + it in the extensions.conf sample, etc. ........ + +2008-12-16 15:00 +0000 [r164602-164623] Russell Bryant <russell@digium.com> + + * apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also, + remove a variable that was not needed. (closes issue #14081) + Reported by: pkempgen + + * /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 + Dec 2008) | 5 lines Don't try to change working directory if a + directory was not configured. (closes issue #14089) Reported by: + caspy ........ + + * channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes + issue #14090) Reported by: alecdavis Patches: + chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license + 585) + +2008-12-16 01:52 +0000 [r164565] Sean Bright <sean.bright@gmail.com> + + * doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also + change a bit of minor formatting. + +2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Open a timer before loading configuration + so that the trunking configuration option will take effect. + (closes issue #14082) Reported by: seandarcy + + * channels/chan_iax2.c: Fix log message to refer to the generic + timing interface, not DAHDI specifically (inspired by issue + #14082) + + * main/frame.c: Make sure we handle a uint32_t payload in + ast_frdup() (closes issue #14080) Reported by: fnordian Patches: + frame.patch uploaded by fnordian (license 110) + +2008-12-15 21:17 +0000 [r164485] Tilghman Lesher <tlesher@digium.com> + + * configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow + disabling pattern match searches within the Realtime dialplan + switch. (closes issue #13698) Reported by: fhackenberger Patches: + 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14) + Tested by: fhackenberger + +2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson <mmichelson@digium.com> + + * apps/app_page.c: Add an 'i' option to app_page. This option works + the same as the 'i' options for app_dial and app_queue, in that + they will ignore any attempts by phones to forward the call. + (closes issue #13977) Reported by: putnopvut Patches: + page_ignore_forwards.patch uploaded by putnopvut (license 60) + Tested by: putnopvut, acunningham + + * /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, + 15 Dec 2008) | 3 lines Add the deadlock note to + ast_spawn_extension as well ........ + + * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged + revisions 164416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec + 2008) | 4 lines Add notes to autoservice and pbx doxygen + regarding a potential deadlock scenario so that it is avoided in + the future ........ + +2008-12-15 19:48 +0000 [r164417] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str + opacity in chan_sip for now, since something wasn't quite right + in the merge. + +2008-12-15 19:42 +0000 [r164415] Steve Murphy <murf@digium.com> + + * include/asterisk/strings.h: I was getting this warning during a + compile on a 64-bit machine running ubuntu server 8.10, and + gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings + being treated as errors In file included from + /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from + chan_vpb.cc:46: + /home/murf/asterisk/trunk/include/asterisk/strings.h: In function + ‘char* ast_str_truncate(ast_str*, ssize_t)’: + /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: + comparison between signed and unsigned integer expressions + make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 + which this fix silences + +2008-12-15 18:12 +0000 [r164351] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 + lines Do not try to unlock a non-existant channel if the transfer + fails. (closes issue #13800) Reported by: dwagner Patches: + asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license + 608) ........ + +2008-12-15 18:09 +0000 [r164349] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c: When querying for the structure of the CDR + table, remove the schema, if it exists. (Closes issue #14058) + +2008-12-15 17:24 +0000 [r164312] Joshua Colp <jcolp@digium.com> + + * main/file.c: Use ast_seekstream to return the file stream back to + the beginning instead of directly seeking to zero. This is + because some audio formats have headers at the front that need to + be skipped, which will be done by the format module. (closes + issue #14079) Reported by: elguero + +2008-12-15 17:21 +0000 [r164272-164309] Russell Bryant <russell@digium.com> + + * channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a + couple more build issues related to ast_str_opaque + + * pbx/pbx_dundi.c: When a reload is issued, always process the + configuration for dundi.conf. The reason is that a reload can be + used to refresh DNS lookups for defined peers. Even if the config + file hasn't changed, we want to process it for that purpose. + (closes issue #13776) Reported by: kombjuder + +2008-12-15 16:16 +0000 [r164268-164270] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix a compile warning and a logic error that + could have been bad for non-realtime queues + + * apps/app_queue.c: Fix up a few issues with regards to queues * + Fix reference counting used in the __queues_show function * Add + code to be sure that the "queue show" command does not print + information for a realtime queue which has been deleted from the + backend * Add a missing unref to the realtime queue loading + function for the case where a queue is in the module's container + but has been deleted from the realtime backend (closes issue + #14033) Reported by: cristiandimache Patches: 14033.patch + uploaded by putnopvut (license 60) Tested by: cristiandimache + +2008-12-15 15:41 +0000 [r164208-164257] Joshua Colp <jcolp@digium.com> + + * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, + configure.ac: Make app_fax compatible with newer versions of + spandsp. This remains backwards compatible with earlier versions + though so do not fret. (closes issue #14073) Reported by: + seandarcy + + * main/utils.c: Update to work with new ast_str changes. + +2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant <russell@digium.com> + + * main/channel.c, /, main/features.c: Merged revisions 164201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) + | 31 lines Handle a case where a call can be bridged to a channel + that is still ringing. The issue that was reported was about a + case where a RINGING channel got redirected to an extension to + pick up a call from parking. Once the parked call got taken out + of parking, it heard silence until the other side answered. + Ideally, the caller that was parked would get a ringing + indication. This patch fixes this case so that the caller + receives ringback once it comes out of parking until the other + side answers. The fixes are: - Make sure we remember that a + channel was an outgoing channel when doing a masquerade. This + prevents an erroneous ast_answer() call on the channel, which + causes a bogus 200 OK to be sent in the case of SIP. - Add some + additional comments to explain related parts of code. - Update + the handling of the ast_channel visible_indication field. Storing + values that are not stateful is pointless. Control frames that + are events or commands should be ignored. - When a bridge first + starts, check to see if the peer channel needs to be given + ringing indication because the calling side is still ringing. - + Rework ast_indicate_data() a bit for the sake of readability. + (closes issue #13747) Reported by: davidw Tested by: russell + Review: http://reviewboard.digium.com/r/90/ ........ + + * apps/app_jack.c: Fix build WRT ast_str_opaque + +2008-12-14 18:16 +0000 [r164168] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/strings.h: Don't pass a negative to an unsigned + type and expect things to work correctly. + +2008-12-14 15:26 +0000 [r164054-164137] Sean Bright <sean.bright@gmail.com> + + * doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art. + + * res/snmp/agent.c: Use ast_str_strlen() instead of recalculating + the string length. + +2008-12-13 13:26 +0000 [r164028] Michiel van Baak <michiel@vanbaak.info> + + * res/snmp/agent.c: nuke another use of the ast_str internals. + +2008-12-13 08:36 +0000 [r163991] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, + funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c, + main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h, + cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c, + res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c, + channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c, + configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c, + main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c, + res/res_config_ldap.c, include/asterisk/threadstorage.h, + cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c, + res/res_config_pgsql.c, main/strings.c (added), main/pbx.c, + channels/chan_sip.c, main/Makefile, main/translate.c, + include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c, + funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c, + main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c, + utils/hashtest2.c, include/asterisk/strings.h, + include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c, + apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque + branch (discontinue usage of ast_str internals) + +2008-12-13 03:03 +0000 [r163951-163952] Sean Bright <sean.bright@gmail.com> + + * doc/tex/asterisk.tex: This shouldn't have gotten commited. We + might want to generate this into a separate file instead of the + version controlled one. + + * doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead + of ASCII art ones. + +2008-12-13 00:59 +0000 [r163912] Joshua Colp <jcolp@digium.com> + + * apps/app_chanspy.c: Only detach and destroy the whisper + audiohooks if they are actually in use. + +2008-12-12 23:48 +0000 [r163873] Terry Wilson <twilson@digium.com> + + * apps/app_queue.c: When using realtime queues, app_queue wasn't + updating the strategy if it was changed in the realtime backend. + This patch resolves the issue for almost all situations. It is + currently not supported to switch to the linear strategy via + realtime since the ao2_container for members will have been set + to have multiple buckets and therefore the members would be + unordered. (closes issue #14034) Reported by: cristiandimache + Tested by: otherwiseguy, cristiandimache + +2008-12-12 23:06 +0000 [r163828] Russell Bryant <russell@digium.com> + + * res/res_clioriginate.c: Add a note to indicate why this only + supports one channel for now. + +2008-12-12 22:04 +0000 [r163762] Tilghman Lesher <tlesher@digium.com> + + * main/editline/read.c, /, main/asterisk.c: Merged revisions 163761 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) + | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, + but also add a pointer inside editline to look back to + asterisk.c, so others don't spend as much time as I did looking + (in the wrong place) for the appropriate function. Reported by: + ZX81, via the #asterisk-users channel Fixed by: me (license 14) + ........ + +2008-12-12 20:12 +0000 [r163716] Russell Bryant <russell@digium.com> + + * CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel + redirect", which is similar in operation to AMI Redirect. Review: + http://reviewboard.digium.com/r/89/ + +2008-12-12 19:16 +0000 [r163675] Steve Murphy <murf@digium.com> + + * channels/chan_dahdi.c: demote always-appearing debug message (for + certain boards) to ast_debug lev 3 msg instead + +2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant <russell@digium.com> + + * main/tcptls.c, channels/chan_sip.c: Rename a number of + tcptls_session variables. There are no functional changes here. + The name "ser" was used in a lot of places. However, it is a + relic from when the struct was a server_instance, not a + session_instance. It was renamed since it represents both a + server or client connection. + + * channels/chan_sip.c: Fix a small race condition in + sip_tcp_locate(). We must increase the reference count on the + tcptls_session _before_ unlocking the thread list. + + * channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with + qualify. The problem was a reference count error on the + tcptls_session structure. (closes issue #13989) Reported by: + Nugget + +2008-12-12 18:17 +0000 [r163629] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: When a device registers we need to unlink + them (if linked) from the peers_by_ip container and link them + back in since their IP address has changed. This would have + manifested itself if you configured a new device (as type=peer), + registered, and then tried to place a call from the device. Since + the peer was not linked into the peers_by_ip container it would + have never been found. (closes issue #13811) Reported by: pj + +2008-12-12 17:22 +0000 [r163582-163612] Michiel van Baak <michiel@vanbaak.info> + + * res/res_monitor.c: Document default Monitor file location. + (closes issue #14065) Reported by: kshumard Patches: + res_monitor.documentation.patch.txt uploaded by kshumard (license + 92) + + * channels/chan_skinny.c: Fix codec capability setup in chan_skinny + Behaviour now is that general codec config flows to default_line + and default_device. [devices] stuff amends default_device and + similar for [lines]. These are copied to individual device and + line as they are created. Added confcapability and confprefs for + the configured stuff which doesn't change as device and so on are + connected. prefs are based on line prefs if they exist, else the + device prefs are used (prefs identifies codec order). (closes + issue #13806) Reported by: pj Patches: codecs.diff uploaded by + wedhorn (license 30) Tested by: pj and me + +2008-12-12 16:55 +0000 [r163579] Joshua Colp <jcolp@digium.com> + + * main/channel.c, channels/chan_sip.c: Since chan_sip is callback + devicestate driven do not pass in actual states, pass in unknown + so we get asked. Additionally do not pass in an actual device + state value in ast_setstate since the channel may be callback + driven. (closes issue #13525) Reported by: pj + +2008-12-12 15:10 +0000 [r163516] Doug Bailey <dbailey@digium.com> + + * configs/phoneprov.conf.sample: Add internationalization to sample + configuration file + +2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) + | 5 lines Specify uint32_t for variables storing a CRC32 so that + it is actually 32 bits on 64-bit machines, as well. (inspired by + issue #13879) ........ + + * main/channel.c, main/autoservice.c, /, + include/asterisk/channel.h: Merged revisions 163448 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 + Dec 2008) | 26 lines Resolve issues that could cause DTMF to be + processed out of order. These changes come from + team/russell/issue_12658 1) Change autoservice to put digits on + the head of the channel's frame readq instead of the tail. If + there were frames on the readq that autoservice had not yet read, + the previous code would have resulted in out of order processing. + This required a new API call to queue a frame to the head of the + queue instead of the tail. 2) Change up the processing of DTMF in + ast_read(). Some of the problems were the result of having two + sources of pending DTMF frames. There was the dtmfq and the more + generic readq. Both were used for pending DTMF in various + scenarios. Simplifying things to only use the frame readq avoids + some of the problems. 3) Fix a bug where a DTMF END frame could + get passed through when it shouldn't have. If code set + END_DTMF_ONLY in the middle of digit emulation, and a digit + arrived before emulation was complete, digits would get processed + out of order. (closes issue #12658) Reported by: dimas Tested by: + russell, file Review: http://reviewboard.digium.com/r/85/ + ........ + +2008-12-11 23:38 +0000 [r163384] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 163383 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) + | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on + certain shells, the terminal is messed up. By intercepting those + events with a signal handler in the remote console, we can avoid + those issues. (closes issue #13464) Reported by: tzafrir Patches: + 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) + Tested by: blitzrage ........ + +2008-12-11 22:49 +0000 [r163317] Matthew Nicholson <mnicholson@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec + 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes + issue #13819) Reported by: adomjan Patches: + pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) + dundi_clearecache3.diff uploaded by mnicholson (license 96) + Tested by: adomjan ........ + +2008-12-11 21:48 +0000 [r163241-163254] Russell Bryant <russell@digium.com> + + * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions + 163253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) + | 8 lines Fix some observed slowdowns in dialplan processing. The + change is to remove autoservice usage from dialplan functions + that do not need it because they do not perform operations that + potentially block. (closes issue #13940) Reported by: tbelder + ........ + + * res/res_timing_pthread.c: Fix a problem where continuous mode + will get inadvertently get turned off if set_rate() is used while + continuous mode was already turned on. (closes issue #13738) + Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix + (license 547) + +2008-12-11 20:57 +0000 [r163198-163213] Mark Michelson <mmichelson@digium.com> + + * configs/voicemail.conf.sample, apps/app_voicemail.c: Add an + option to voicemail.conf to allow urgent messages to be forwarded + as not urgent. (closes issue #14063) Reported by: jaroth Patches: + urgfwd_v2.patch uploaded by jaroth (license 50) + + * main/features.c: Add an appropriate goto if ast_call fails + +2008-12-11 20:07 +0000 [r163171] Russell Bryant <russell@digium.com> + + * main/channel.c: Fix the "failed" extension for outgoing calls. + The conversion to use ast_check_hangup() everywhere instead of + checking the softhangup flag directly introduced this problem. + The issue is that ast_check_hangup() checked for tech_pvt to be + NULL. Unfortunately, this will be NULL is some valid + circumstances, such as with a dummy channel. The fix is simple. + Don't check tech_pvt. It's pointless, because the code path that + sets this to NULL is when the channel hangup callback gets + called. This happens inside of ast_hangup(), which is the same + function responsible for freeing the channel. Any code calling + ast_check_hangup() better not be calling it after that point, and + if so, we have a bigger problem at hand. (closes issue #14035) + Reported by: erogoza + +2008-12-11 20:02 +0000 [r163168] Tilghman Lesher <tlesher@digium.com> + + * configure, configure.ac: Sometimes even Linux needs -lm to link + libtonezone, such as when libtonezone is compiled statically. + (closes issue #13887) Reported by: tzafrir + +2008-12-11 19:40 +0000 [r163166] Mark Michelson <mmichelson@digium.com> + + * main/features.c: Reduce indentation level of + ast_feature_request_and_dial + +2008-12-11 17:06 +0000 [r163094] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 163092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) + | 11 lines Fix an issue that made it so you could only have a + single caller executing a custom feature at a time. This was + especially problematic when custom features ran for any + appreciable amount of time. The fix turned out to be quite + simple. The dynamic features are now stored in a read/write list + instead of a list using a mutex. (closes issue #13478) Reported + by: neutrino88 Fix suggested by file ........ + +2008-12-11 16:52 +0000 [r163089] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 163088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) + | 6 lines Don't wait forever, if there's a specified recording + timeout. (closes issue #13885) Reported by: bamby Patches: + res_agi.c.patch uploaded by bamby (license 430) ........ + +2008-12-11 16:47 +0000 [r163081-163085] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 163084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec + 2008) | 4 lines Revert this cast to long. Using time_t here + causes build failures on a FreeBSD 32-bit build. ........ + + * /, apps/app_queue.c: Merged revisions 163080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec + 2008) | 14 lines Fix a potential crash due to unsafe datastore + handling. This patch also contains a conversion from using long + to time_t for representing times for a queue, as well as some + whitespace fixes. (closes issue #14060) Reported by: nivek + Patches: datastore_fixup.patch.corrected uploaded by nivek + (license 636) with slight modification from me Tested by: nivek + ........ + +2008-12-11 15:40 +0000 [r163037] Sean Bright <sean.bright@gmail.com> + + * doc/tex/qos.tex: Fix some of the grammar issues in + doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard + Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92) + (Slight modifications by seanbright) + +2008-12-11 15:05 +0000 [r162997] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: When a device registers to use it is + entirely possible that they may be in use, so tell the core that + we don't know the devstate and have it ask us for it. (closes + issue #13525) Reported by: pj + +2008-12-10 23:01 +0000 [r162930] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Previously missing line, now the substitution works + correctly + +2008-12-10 22:53 +0000 [r162927] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 + Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check. + Pointed out by mmichelson, thanks! ........ + +2008-12-10 22:48 +0000 [r162923] Joshua Colp <jcolp@digium.com> + + * res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to + changes done to turn it into a single memory allocation we can't + just use the existing CLI alias structure. We have to destroy all + existing ones and then create new ones. (closes issue #14054) + Reported by: pj + +2008-12-10 22:48 +0000 [r162922] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Checking global variables here actually overwrote the + previous substitution by channel variables, and in any case, was + redundant; pbx_substitute_variables_helper ALREADY does + substitution for global variables. (closes issue #13327) Reported + by: pj + +2008-12-10 22:11 +0000 [r162891] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 + Dec 2008) | 5 lines (closes issue #13229) Reported by: + clegall_proformatique Ensure that moh_generate does not return + prematurely before local_ast_moh_stop is called. Also, the sleep + in mp3_spawn now only occurs for http locations since it seems to + have been added originally only for failing media streams. + ........ + +2008-12-10 19:02 +0000 [r162739-162805] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 + lines Fix subscription based MWI up a bit. We only want to put + sip: at the beginning of the URI if it is not already there and + revert code to ignore destination check if subscribing for MWI. + (closes issue #12560) Reported by: vsauer Patches: patch001.diff + uploaded by ramonpeek (license 266) ........ + + * /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 + lines When a SIP peer unregisters set the expiry time back to 0 + so that the 200 OK contains an expires of 0. (closes issue + #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded + by hjourdain (license 583) ........ + +2008-12-10 17:09 +0000 [r162687] Michiel van Baak <michiel@vanbaak.info> + + * include/asterisk.h, main/asterisk.c, main/cli.c: add tab + completion for 'core set debug X filename.c' (closes issue + #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt + uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel + +2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson <mmichelson@digium.com> + + * doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec + 2008) | 8 lines Add missing documentation to misdn.txt (closes + issue #14052) Reported by: festr Patches: misdn.txt.patch + uploaded by festr (license 443) ........ + + * /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec + 2008) | 11 lines Revert fix for issue 13570. It has caused more + problems than it helped to fix. (closes issue #13783) Reported + by: navkumar (closes issue #14025) Reported by: ffs ........ + +2008-12-10 16:11 +0000 [r162619-162660] Joshua Colp <jcolp@digium.com> + + * res/res_http_post.c: FreeBSD also needs libgen.h (closes issue + #14051) Reported by: ys Patches: res_http_post.c.diff uploaded by + ys (license 281) + + * /, main/rtp.c: Merged revisions 162653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 + lines Increment the sequence number on the end packets for + RFC2833. After reading the RFC some more and doing some testing I + agree with this change. (closes issue #12983) Reported by: vt + Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license + 520) ........ + + * channels/chan_sip.c: When transmitting a register set the socket + port to the local one for the transport being used, not the port + for the remote server. (closes issue #13633) Reported by: + performer + +2008-12-10 11:34 +0000 [r162583] Michiel van Baak <michiel@vanbaak.info> + + * res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD + uses an old version of gcc which throws an error if you use a + macro that's not #defined + +2008-12-10 01:09 +0000 [r162542] Joshua Colp <jcolp@digium.com> + + * doc/janitor-projects.txt, channels/iax2-parser.c, + apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and + remove it as a janitor project. (closes issue #14032) Reported + by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) + +2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/stringfields.h: it does help if the compiler + attribute syntax is correct + +2008-12-09 23:10 +0000 [r162466] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 + Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ + +2008-12-09 22:38 +0000 [r162414-162418] Russell Bryant <russell@digium.com> + + * include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen, + main/asterisk.c: Add some additional Asterisk project developer + documentation. After the nightly update of the documentation on + asterisk.org, I'll post an update to asterisk-dev with a pointer + to the changes. This covers some release branch and commit policy + information. None of this should be a surprise, since it's just + documenting what we have already been doing. + + * include/asterisk/utils.h, /, main/utils.c, main/asterisk.c: + Merged revisions 162413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) + | 8 lines Remove the test_for_thread_safety() function + completely. The test is not valid. Besides, if we actually + suspected that recursive mutexes were not working, we would get a + ton of LOG_ERROR messages when DEBUG_THREADS is turned on. + (inspired by a discussion on the asterisk-dev list) ........ + +2008-12-09 21:57 +0000 [r162355] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 + Dec 2008) | 4 lines We appear to have documented tz= in the + [general] section of voicemail.conf, without actually having + implemented it. Oops. (Reported by Olivier on the -users list) + ........ + +2008-12-09 21:16 +0000 [r162342] Joshua Colp <jcolp@digium.com> + + * /, apps/app_directed_pickup.c: Merged revisions 162341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 + lines Add 'down' as a valid state for directed call pickup. This + creeps up when we receive session progress when dialing a device + and not ringing. (closes issue #14005) Reported by: ddl ........ + +2008-12-09 20:59 +0000 [r162291] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) + | 9 lines Fix an issue where callers on an incoming call on an + SLA trunk would not hear ringback. We need to make sure that we + don't start writing audio to the trunk channel until we're + actually ready to answer it. Otherwise, the channel driver will + treat it as inband progress, even though all they are getting is + silence. (closes issue #12471) Reported by: mthomasslo ........ + +2008-12-09 20:46 +0000 [r162275] Joshua Colp <jcolp@digium.com> + + * /, apps/app_festival.c: Merged revisions 162273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 + lines Fix double declaration of 'x' on the PPC platform. (closes + issue #14038) Reported by: ffloimair ........ + +2008-12-09 20:40 +0000 [r162271] Steve Murphy <murf@digium.com> + + * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 + line In discussion with seanbright on #asterisk-dev, I have added + a default rule, and an option to suppress the default rule from + being generated in the flex output, for the sake of those OS's + where they didn't tweak flex's ECHO macro, and the compiler + doesn't like it. The regressions are OK with this. ........ + +2008-12-09 20:30 +0000 [r162266] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, /: Merged revisions 162265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec + 2008) | 6 lines If we fail to start a thread for the pbx to run + in, we need to be sure to decrease the number of active calls on + the system. This fix may relate to ABE-1713, but it is not + certain yet. ........ + +2008-12-09 19:48 +0000 [r162197-162205] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 162204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 + lines Make sure that the timestamp for DTMF is not the same as + the previous voice frame and do not send audio when transmitting + DTMF as this confuses some equipment. (closes issue #13209) + Reported by: ip-rob Patches: 13209.diff uploaded by file (license + 11) Tested by: ip-rob, bujones ........ + + * /, main/rtp.c: Merged revisions 162188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 + lines Take video into account when early bridging RTP. (closes + issue #13535) Reported by: davidw ........ + +2008-12-09 18:35 +0000 [r162079-162140] Steve Murphy <murf@digium.com> + + * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 + line Previous fix used ast_malloc and ast_copy_string and messed + up the standalone stuff. Fixed. ........ + + * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c, + res/ael/ael.flex: Merged revisions 162013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | + 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: + 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, + murf This crash was the result of a few small errors that would + combine in 64-bit land to result in a crash. 32-bit land might + have seen these combine to mysteriously drop the args to an + application call, in certain circumstances. Also, in trying to + find this bug, I spotted a situation in the flex input, where, in + passing back a 'word' to the parser, it would allocate a buffer + larger than necessary. I changed the usage in such situations, so + that strdup was not used, but rather, an ast_malloc, followed by + ast_copy_string. I removed a field from the pval struct, in u2, + that was never getting used, and set in one spot in the code. I + believe it was an artifact of a previous fix to make switch cases + work invisibly with extens. And, for goto's I removed a '!' from + before a strcmp, that has been there since the initial merging of + AEL2, that might prevent the proper target of a goto from being + found. This was pretty harmless on its own, as it would just + louse up a consistency check for users. Many thanks to + ckjohnsonme for providing a simplified and complete set of + information about the bug, that helped considerably in finding + and fixing the problem. Now, to get aelparse up and running again + in trunk, and out of its "horribly broken" state, so I can run + the regression suite! ........ + +2008-12-09 16:47 +0000 [r161951-162016] Russell Bryant <russell@digium.com> + + * /, apps/app_disa.c: Merged revisions 162014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) + | 5 lines Allow DISA to handle extensions that start with #. + (closes issue #13330) Reported by: jcovert ........ + + * /, main/app.c: Merged revisions 161948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) + | 15 lines Fix a problem with GROUP() settings on a masquerade. + The previous code carried over group settings from the old + channel to the new one. However, it did nothing with the group + settings that were already on the new channel. This patch removes + all group settings that already existed on the new channel. I + have a more complicated version of this patch which addresses + only the most blatant problem with this, which is that a channel + can end up with multiple group settings in the same category. + However, I could not think of a use case for keeping any of the + group settings from the old channel, so I went this route for + now. (closes AST-152) ........ + +2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons <eliels@gmail.com> + + * funcs/func_odbc.c: Avoid allocating memory for a thread that + don't need it. Also, this memory was not being freed until the + main thread ends. (That is never). (closes issue #14040) Reported + by: eliel Patches: func_odbc.c.patch uploaded by eliel (license + 64) + +2008-12-08 23:04 +0000 [r161911] Brandon Kruse <bkruse@digium.com> + + * main/pbx.c: Note that the recently changed waittime parameter is + in milliseconds. + +2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp <jcolp@digium.com> + + * formats/format_pcm.c: Add alw as a valid file extension for alaw + and ulw as a valid file extension for ulaw. (closes issue #14001) + Reported by: henrikw Patches: alw.diff uploaded by henrikw + (license 627) + + * contrib/scripts/autosupport.8, contrib/scripts/autosupport: + Update autosupport script with a few changes. + +2008-12-08 18:49 +0000 [r161790] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c: Allocate enough space initially for the message. + (closes issue #14027) Reported by: junky Patches: M14027.diff + uploaded by junky (license 177) + +2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp <jcolp@digium.com> + + * main/pbx.c: Fix a regression introduced when the PBX timeouts + were converted to milliseconds. collect_digits now gets + milliseconds fed to it, not seconds. (closes issue #14012) + Reported by: dveiga Patches: 14012.patch uploaded by bkruse + (license 132) + + * /, channels/chan_sip.c: Merged revisions 161725 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 + lines Make the usereqphone option work again. (closes issue + #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff + uploaded by mmaguire (license 571) ........ + +2008-12-08 17:23 +0000 [r161721] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c: Fix a crash that can occur on a transfer in + chan_sip when attempting to collect rtp stats. (closes issue + #13956) Reported by: chris-mac Tested by: chris-mac + +2008-12-08 16:02 +0000 [r161679] Terry Wilson <twilson@digium.com> + + * channels/chan_sip.c, CHANGES: Add the ability to play a courtesy + tone to the transfer target in a native SIP attended transfer by + setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. + +2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons <eliels@gmail.com> + + * main/xmldoc.c: - Fix a leak while printing an argument + description. - Avoid printing the name of an argument in the + [Arguments] tag if there is no description for that argument. + + * apps/app_voicemail.c: Add voicemail related applications and + functions XML documentation: applications: - VoiceMail() - + VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - + MAILBOX_EXISTS() + + * apps/app_sms.c: Introduce SMS() application XML documentation. + +2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons <eliels@gmail.com> + + * apps/app_speech_utils.c: Move Speech* applications and functions + documentation to XML. + +2008-12-05 23:24 +0000 [r161493] Mark Michelson <mmichelson@digium.com> + + * apps/app_stack.c: If the autoloop flag is set on a channel, then + we need to add 1 to the priority when checking if the extension + exists. Otherwise, gosubs will fail. This was discovered when + investigating an asterisk-users mailing list post made by Gary + Hawkins. + +2008-12-05 21:08 +0000 [r161349-161427] Sean Bright <sean.bright@gmail.com> + + * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions + 161426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 + (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec + 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned + int). (closes issue #14006) Reported by: alphaque Patches: + astobj2.h-patch uploaded by alphaque (license 259) (Slightly + modified by seanbright) ........ ................ + + * apps/app_voicemail.c: Use ast_free() instead of free(), pointed + out by eliel on IRC. + + * apps/app_voicemail.c: When using IMAP_STORAGE, it's important to + convert bare newlines (\n) in emailbody and pagerbody to CR-LF so + that the IMAP server doesn't spit out an error. This was + informally reported on #asterisk-dev a few weeks ago. Reviewed by + Mark M. on IRC. + +2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 161287 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) + | 2 lines Fix a NULL format string warning found by buildbot. + ........ + + * apps/app_minivm.c: Resolve a compiler warning from buildbot about + a NULL format string. + +2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons <eliels@gmail.com> + + * main/udptl.c, main/frame.c, res/res_musiconhold.c, + channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c, + main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c, + channels/chan_skinny.c, res/res_agi.c, main/features.c, + apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c, + res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c, + apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c, + channels/chan_sip.c, main/translate.c, channels/chan_agent.c, + res/res_convert.c, res/res_crypto.c, apps/app_queue.c, + channels/chan_oss.c, apps/app_playback.c, + channels/chan_usbradio.c, main/file.c, main/astmm.c, + pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c, + apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c, + apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible. + (closes issue #13990) Reported by: eliel Patches: array_len.diff + uploaded by eliel (license 64) + +2008-12-05 05:41 +0000 [r161181] Tilghman Lesher <tlesher@digium.com> + + * main/config.c: The first file should have a blank config filename + in the structure, so that when a save occurs to a different + filename, everything goes to the alternate filename, instead of + appending to the original. This is important for the AMI command + UpdateConfig. (closes issue #13301) Reported by: trevo Patches: + 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14) + 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license + 14) Tested by: Corydon76, blitzrage + +2008-12-05 02:47 +0000 [r161147] Sean Bright <sean.bright@gmail.com> + + * apps/app_voicemail.c: Check the return value of fread/fwrite so + the compiler doesn't complain. Only a problem when IMAP_STORAGE + is enabled. + +2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If + 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it + exists) after T38 is negotiated. Terry Wilson created the + original patch for this functionality, which I slightly modified + and added the faxdetect=yes|no configuration option. This patch + is only for T38 fax detection and does not do anything for G711 + over SIP fax detection. By default, this option is disabled. + Reviewboard: http://reviewboard.digium.com/r/69/ This + functionality is for issue AST-140. + +2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons <eliels@gmail.com> + + * main/cli.c: Fix minor coding guidelines introduced with CLI + permissions. + +2008-12-04 18:32 +0000 [r161014] Jeff Peeler <jpeeler@digium.com> + + * /, main/rtp.c: Merged revisions 161013 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) + | 9 lines (closes issue #13835) Reported by: matt_b Tested by: + jpeeler This mirrors a check that was present in ast_rtp_read to + also be in ast_rtp_raw_write to not schedule sending the receiver + report if the remote RTCP endpoint address isn't present in the + RTCP structure. Closes AST-142. ........ + +2008-12-04 16:45 +0000 [r160945] Mark Michelson <mmichelson@digium.com> + + * /, main/callerid.c: Merged revisions 160943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec + 2008) | 15 lines Fix a callerid parsing issue. If someone + formatted callerid like the following: "name <number>" (including + the quotation marks), then the parts would be parsed as name: + "name number: number This is because the closing quotation mark + was not discovered since the number and everything after was + parsed out of the string earlier. Now, there is a check to see if + the closing quote occurs after the number, so that we can know if + we should strip off the opening quote on the name. Closes AST-158 + ........ + +2008-12-04 16:37 +0000 [r160938] Michiel van Baak <michiel@vanbaak.info> + + * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug + flag so skinny debug will show information about packets. We dont + want to scare users with this, so we added a devmode compile flag + (closes issue #13952) Reported by: wedhorn Patches: + packetdebug3.diff uploaded by wedhorn (license 30) Tested by: + mvanbaak, wedhorn + +2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons <eliels@gmail.com> + + * res/res_agi.c: Added XML documentation for the following AGI + commands: - get option - get variable - hangup - noop + +2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett <rmudgett@digium.com> + + * funcs/func_callerid.c: Jcolp pointed out that num will also match + number + + * funcs/func_callerid.c: * Found a couple more places where + num/number needed to be done so 1.4 upgraders will not have + problems. * Added curly braces and minor tweaks. + +2008-12-03 21:58 +0000 [r160791] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 160770 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 + Dec 2008) | 2 lines Some compilers warn on null format strings; + some don't (caught by buildbot) ........ + +2008-12-03 21:09 +0000 [r160760] Steve Murphy <murf@digium.com> + + * /, funcs/func_callerid.c: Merged revisions 160703 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec + 2008) | 11 lines (closes issue #13597) Reported by: john8675309 + Patches: patch.13597 uploaded by murf (license 17) Tested by: + murf, john8675309 This patch causes the setcid func to update the + CDR clid after setting the channel field. I also notice that in + trunk, the num/number of 1.4 is left out; I decided to include + the option to use either in trunk, so as not to have 1.4 + upgraders not to have problems. ........ + +2008-12-03 20:35 +0000 [r160699-160700] Jason Parker <jparker@digium.com> + + * main/manager.c: Another place this is missing + + * main/manager.c: Fix typo when ListCategories returns none. + (closes issue #13994) Reported by: mika Patches: + ListCategoriesActionPatch.diff uploaded by mika (license 624) + +2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons <eliels@gmail.com> + + * channels/iax2-provision.c: - iax2-provision was not freeing + iax_templates structure when unloading the chan_iax2.so module. - + Move the code to start using the LIST macros. Review: + http://reviewboard.digium.com/r/72 (closes issue #13232) Reported + by: eliel Patches: iax2-provision.patch.txt uploaded by eliel + (license 64) (with minor changes pointed by Mark Michelson on + review board) Tested by: eliel + +2008-12-03 18:37 +0000 [r160626] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: Add some + safety measures when using gosub, especially when using the + options for app_dial and app_queue to run a gosub when the call + is answered. * Check for the existence of the gosub target in + gosub_exec. If it is nonexistent, then this will cause errors + when we attempt to actually run the gosub, including a definite + memory leak and potential crashes. Return an error in this + situation * Check the return value of pbx_exec in app_dial and + app_queue before attempting to actually run the gosub routine. If + there was an error, we should not attempt to run the gosub. * + Change a '|' to a ',' in app_queue. * Add some extra curly braces + where they had been missing previously. (closes issue #13548) + Reported by: fiddur + +2008-12-03 17:48 +0000 [r160562] Eliel C. Sardanons <eliels@gmail.com> + + * apps/app_minivm.c: - Add <variable /> tags when naming a channel + variable. - Add <filename /> tags when naming a filename. - + Simplify the xml formatting putting some enters. + +2008-12-03 17:38 +0000 [r160559] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 160558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) + | 7 lines If an entry is added to the directory during a scan + when another entry expires, then that new entry will not be + processed promptly, but must wait for either a future entry to + start or a current entry's retry to occur. If no other entries + exist in the directory (other than the new entries) when a bunch + expire, then the new entries must wait until another new entry is + added to be processed. This was a rather weird race condition, + really. Fixes AST-147. ........ + +2008-12-03 17:07 +0000 [r160555] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: When investigating issue #13548, I found that + gosub handling in app_queue was just completely wrong, mostly + because the channel operations being performed were being done on + the incorrect channel. With this set of changes, a gosub will + correctly run on the answering queue member's channel. There are + still crash issues which occur if there are dialplan syntax + errors, so I cannot yet close the referenced issue. + +2008-12-03 17:01 +0000 [r160481-160552] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 160551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) + | 4 lines Don't start scanning the directory until all modules + are loaded, because some required modules (channels, apps, + functions) may not yet be in memory yet. Fixes AST-149. ........ + + * /, channels/chan_sip.c: 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 11:01 +0000 [r160447] Eliel C. Sardanons <eliels@gmail.com> + + * apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are + using XML documentation (or the xml documentation wont be + loaded). - Use <variable></variable> to refer to a dialplan + variable. + +2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Info on LOCAL_PEEK function. + + * apps/app_stack.c: Add LOCAL_PEEK function, as requested by + lmadsen. + +2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: remove duplicate comment that I + accidentally merged + + * channels/chan_dahdi.c: (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 17:56 +0000 [r160208-160308] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: 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) ........ + + * main/pbx.c, main/frame.c, /, channels/chan_features.c, + include/asterisk/stringfields.h, apps/app_voicemail.c, + main/cli.c: 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 23:37 +0000 [r160170-160172] Sean Bright <sean.bright@gmail.com> + + * main/manager.c, /: 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 ........ + + * channels/chan_phone.c: Silence a build warning. + (chan_phone.c:810: warning: value computed is not used) + + * utils/smsq.c: Pay attention to the return value of system(), even + if we basically ignore it. + +2008-12-01 21:23 +0000 [r160097] Tilghman Lesher <tlesher@digium.com> + + * configure, configure.ac: Use AST_EXT_LIB_SETUP before using + AST_EXT_LIB_CHECK or bad things happen. + +2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons <eliels@gmail.com> + + * configs/cli_permissions.conf.sample (added), configure, + include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/cli.h, include/asterisk/_private.h, CHANGES, + main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on + cli_permissions.conf configuration file, we are able to permit or + deny cli commands based on some patterns and the local user and + group running rasterisk. (Sorry if I missed some of the testers). + Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue + #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, + otherwiseguy, mvanbaak + +2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: 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. + ........ + + * /, .cleancount: 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 ... + ........ + +2008-12-01 14:09 +0000 [r159898] Michiel van Baak <michiel@vanbaak.info> + + * main/manager.c, /: 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. ........ + +2008-11-29 18:33 +0000 [r159853] Tilghman Lesher <tlesher@digium.com> + + * apps/app_readexten.c: Allow the '#' sign to exist within an + extension (inspired by issue #13330) + +2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c, + utils/frame.c, include/asterisk/astmm.h, configure, + include/asterisk/compat.h, main/features.c, + include/asterisk/module.h, main/logger.c, + include/asterisk/dlinkedlists.h, main/dns.c, + include/asterisk/utils.h, include/asterisk/devicestate.h, + channels/chan_sip.c, include/asterisk/dundi.h, + include/asterisk/enum.h, configure.ac, channels/chan_agent.c, + include/asterisk/config.h, utils/astman.c, + include/asterisk/cli.h, include/asterisk/channel.h, + include/jitterbuf.h, include/asterisk/manager.h, + utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, + include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h, + main/srv.c, channels/chan_misdn.c, + include/asterisk/linkedlists.h, main/event.c, + include/asterisk/lock.h, include/asterisk/strings.h, + utils/extconf.c, makeopts.in, include/asterisk/stringfields.h, + main/xmldoc.c, utils/check_expr.c: 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 + + * Makefile, funcs/func_sprintf.c (added), main/Makefile, + channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt, + res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile: + we can now build with -Wformat=2, which found a couple of real + bugs because SPRINTF() use non-literal format strings (which + cannot be checked), move it into its own module so the rest of + func_strings can benefit from format string checking + +2008-11-28 14:20 +0000 [r159734] Michiel van Baak <michiel@vanbaak.info> + + * res/Makefile: Make res_config_ldap compile with the official + OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define + from their source and since we are using a couple of deprecated + function calls we should define it with a CFLAG. Tested by me on + OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps + compiling. It shouldn't break, we only define the LDAP_DEPRECATED + with this which is what all 2.2.X and older versions of OpenLDAP + did in their own tree. + +2008-11-27 20:29 +0000 [r159701] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Removed duplicate code + +2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant <russell@digium.com> + + * main/pbx.c: Make a formatting change to test a new post-commit + hook for reviewboard. http://reviewboard.digium.com/r/65/ + + * main/pbx.c: Make a formatting change to test a new post-commit + hook for reviewboard. http://reviewboard.digium.com/r/65/ + + * main/pbx.c: Make a formatting change to test a new post-commit + hook for reviewboard. http://reviewboard.digium.com/r/65/ + +2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/agi.h, configure, + include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen, + autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c, + apps/app_stack.c, include/asterisk/optional_api.h (added): + improve handling of API calls provided by loaded modules through + use of some GCC features; this makes app_stack's usage of AGI + APIs even cleaner, and will allow it to work 'as expected' either + with or without res_agi being loaded reviewed at + http://reviewboard.digium.com/r/62 + + * main/manager.c, CHANGES: add support for event suppression for + AMI-over-HTTP + +2008-11-26 19:57 +0000 [r159554] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c: 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 19:08 +0000 [r159534] Kevin P. Fleming <kpfleming@digium.com> + + * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, + Makefile.rules: 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 18:33 +0000 [r159475] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c: 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 14:58 +0000 [r159437] Mark Michelson <mmichelson@digium.com> + + * channels/chan_agent.c: 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 + +2008-11-26 03:18 +0000 [r159402] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: Always parse arguments in park_call_exec so that + app_args is valid. This prevents a crash when executing Park from + the dialplan with no arguments. + +2008-11-25 23:03 +0000 [r159360] Steve Murphy <murf@digium.com> + + * main/cdr.c, /, channels/chan_iax2.c: 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 22:45 +0000 [r159276-159317] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option, + waitfordialtone, for UK analog lines which do not end a call + until the originating line hangs up. (closes issue #12382) + Reported by: one47 Patches: + zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license + 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed + (license 463) Tested by: fleed + + * /, channels/chan_iax2.c: 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 ........ + +2008-11-25 21:49 +0000 [r159250] Mark Michelson <mmichelson@digium.com> + + * apps/app_followme.c: Make the options for the general and + profiles more consistent for the "pls_hold_prompt" option. This + does not affect any released version of Asterisk, so there is no + need to update the CHANGES file for this. (closes issue #13893) + Reported by: eliel + +2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: 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 ........ ................ + + * pbx/pbx_realtime.c: Don't actually do anything with a negative + priority, because we ignore it in the result, anyway. + + * main/pbx.c: Don't limit the length of the hint at the final step + (from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80 + chars max). This will allow more channels to be used in a single + hint. + +2008-11-25 16:18 +0000 [r159093] Terry Wilson <twilson@digium.com> + + * apps/app_festival.c: Add missing variable declaration for PPC + code + +2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher <tlesher@digium.com> + + * apps/app_readexten.c: Copyright clarification; also, have + variable set to "t" or "i" on timeout or invalid extension, + respectively. (closes issue #13944) Reported by: chappell + + * channels/chan_usbradio.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/xpmr/xpmr.c, apps/app_rpt.c: Merged revisions 159025 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) + | 3 lines System call ioperm is non-portable, so check for its + existence in autoconf. (Closes issue #13863) ........ + +2008-11-25 03:49 +0000 [r158992] Terry Wilson <twilson@digium.com> + + * channels/chan_usbradio.c: Make chan_usbradio compile under dev + mode + +2008-11-25 01:01 +0000 [r158959] Sean Bright <sean.bright@gmail.com> + + * funcs/func_dialgroup.c, channels/chan_sip.c, + include/asterisk/astobj2.h, res/res_phoneprov.c, + main/taskprocessor.c, channels/chan_console.c, + channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c, + main/config.c, main/manager.c, res/res_timing_pthread.c, + main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c, + res/res_clialiases.c: This is basically a complete rollback of + r155401, as it was determined that it would be best to maintain + API compatibility. Instead, this commit introduces + ao2_callback_data() which is functionally identical to + ao2_callback() except that it allows you to pass arbitrary data + to the callback. Reviewed by Mark Michelson via ReviewBoard: + http://reviewboard.digium.com/r/64 + +2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson <mnicholson@digium.com> + + * main/file.c: Fix compiling in dev mode. + + * UPGRADE.txt, apps/app_queue.c: Make the Join event from app_queue + use CallerIDNum insead of CallerID for indicating the callerid + number just like the rest of asterisk. (closes issue #13883) + Reported by: davidw + + * main/manager.c, res/res_agi.c, include/asterisk/manager.h: Added + EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes + issue #13873) Reported by: fnordian Patches: ami_agievent.patch + uploaded by fnordian (license 110) + +2008-11-24 21:52 +0000 [r158857] Tilghman Lesher <tlesher@digium.com> + + * main/dsp.c: Add a bit of documentation (thanks, I-MOD) on what + the silence threshold constant actually does and what values are + valid for it. + +2008-11-24 21:27 +0000 [r158851] Matthew Nicholson <mnicholson@digium.com> + + * main/file.c: Make ast_streamfile() check the result of + ast_openstream() before doing anything with it. (closes issue + #13955) Reported by: chris-mac + +2008-11-24 18:11 +0000 [r158808] Terry Wilson <twilson@digium.com> + + * apps/app_minivm.c: This patch adds a new application for sending + MWI to phones via Asterisk's event subsystem. Also, the minivm + documentation is all converted to use xmldocs. (closes issue + #13946) Reported by: Marquis Patches: + minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32) + Tested by: otherwiseguy, Marquis + +2008-11-23 03:36 +0000 [r158754-158756] Sean Bright <sean.bright@gmail.com> + + * channels/chan_sip.c, configs/sip.conf.sample: If you enabled + 'notifycid' one of the limitations is that the calling channel is + only found if it dialed the extension that was subscribed to. You + can now specify 'ignore-context' for the 'notifycid' option in + sip.conf which will, as it's value implies, ignore the current + context of the caller when doing the lookup. + + * channels/chan_sip.c: No need to use a separate structure for this + since we can just pass our sip_pvt pointer in directly. + +2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak <michiel@vanbaak.info> + + * funcs/func_realtime.c: last commit worked on OpenBSD but still + generated warning on Ubuntu. Initialise a variable so + --enable-dev-mode does not complain + + * channels/chan_skinny.c: dont send reorder tone after a device is + hungup if a dialout is abandoned or failed. Without this reorder + tone will play after hangup and both wedhorn's and my wife have + threatened to use an axe on our asterisk box (closes issue + #13948) Reported by: wedhorn Patches: switch.diff uploaded by + wedhorn (license 30) + + * channels/chan_skinny.c: Add Media Source Update to skinny's + control2str (issue #13948) + + * channels/chan_skinny.c: fix a very occasional core dump in + chan_skinny found by wedhorn. (issue #13948) + + * funcs/func_realtime.c: make this compile under devmode + +2008-11-21 23:40 +0000 [r158606] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 158603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | + 11 lines In reference to the fix made for 13871, I was merging + the fix into 1.6.0 and realized I missed the code in the h-exten + block, and didn't catch it because my test case had the h-exten + commented out. So, this corrects the code I missed, as a + preventative against another crash report. Tested with the + h-exten defined, all is well. ........ + +2008-11-21 23:33 +0000 [r158602-158605] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Allow space within an extension, when the space is + within a character class. (requested by lmadsen on -dev, patch by + me) + + * main/pbx.c, /: 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) ........ + +2008-11-21 22:12 +0000 [r158540] Russell Bryant <russell@digium.com> + + * /, include/asterisk/astobj2.h, main/astobj2.c: 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 21:47 +0000 [r158484] Steve Murphy <murf@digium.com> + + * /, main/features.c: 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 21:06 +0000 [r158482] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c: Fix for #13963. Make physical channel + mapping unconfigured default + +2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming <kpfleming@digium.com> + + * UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt, + CHANGES: as suggested by jtodd, document the purposes of the + CHANGES and UPGRADE files + +2008-11-21 19:40 +0000 [r158414] Jason Parker <jparker@digium.com> + + * main/manager.c: Make sure we add the Event header for + CoreShowChannels. (closes issue #13334) Reported by: srt Patches: + 13334_missing_event_header_in_core_show_channel.diff uploaded by + srt (license 378) + +2008-11-21 17:08 +0000 [r158374] Terry Wilson <twilson@digium.com> + + * cdr/cdr_csv.c: 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 15:53 +0000 [r158315] Doug Bai |