diff options
| author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-08-23 18:36:15 +0000 |
|---|---|---|
| committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-08-23 18:36:15 +0000 |
| commit | 4a85f53b89164e250bd93142410eff58af0d4c4c (patch) | |
| tree | 0244bbb0d18c7c9581f48a07c6ab410708baa9bd | |
| parent | 2603941ac7e28e2f02a29aa48d8ca21f4306f550 (diff) | |
Importing files for 1.6.2.12-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.12-rc1@283280 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 26846 |
3 files changed, 26848 insertions, 0 deletions
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..5617af91e --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.2.12-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..c9e63119f --- /dev/null +++ b/ChangeLog @@ -0,0 +1,26846 @@ +2010-08-23 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.12-rc1 Released. + +2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500 + (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from + https://origsvn.digium.com/svn/asterisk/trunk .......... r278274 + | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 + line Reference correct struct member for unlikely event + PRI_EVENT_CONFIG_ERR. .......... ................ + + * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 + Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending + ALERTING is a protocol error The PRI layer in chan_dadhi will + check if a PROGRESS message has already been sent, and not allow + sending another (although that is technically allowed by the Q931 + spec), however it does not protect against sending an ALERTING + and then sending a PROGRESS message, which is a violation of the + specification. Most switches don't seem to care too deeply about + this, but some do, and will disconnect the call when receiving + this invalid sequence. Protocol specification reference: + T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview + protocol control (network side) point-point (sheet 3 of 8)" + (closes issue #17874) Reported by: nic_bellamy Patches: + asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by + nic bellamy (license 299) + asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) + asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) ........ + +2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) + | 11 lines tos_sip option was not being set correctly When + tos_sip is used, the tos of the sip socket is only set correctly + if the socket binding changes on a reload. If the binding stays + the same but the TOS changes, the new tos value would not take + into effect. This patch fixes that. (closes issue #17712) + Reported by: nickb ........ + + * channels/chan_sip.c: fixes sip peer memory leaks in the + peer_by_ip table (issue #17798) + +2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c: Merged revisions 277944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul + 2010) | 16 lines Regression with T.38 negotiation Prior to + 1.4.26.3 T.38 negotiation worked properly, in the case of the + reporter. (issue #16852) Reported by: cfc (closes issue #16705) + Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded + by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, + samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ + +2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson@digium.com> + + * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 + Aug 2010) | 2 lines Add some documentation about codec + negotiation to sip.conf ........ + +2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes crash with notifycid (closes issue + #17868) Reported by: francesco_r Patches: issue_17868.diff + uploaded by dvossel (license 671) Tested by: francesco_r + +2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c: Don't warn on callerid when completely + text, instead of numeric with localdialplan prefixes. (closes + issue #16770) Reported by: jamicque Patches: + 20100413__issue16770.diff.txt uploaded by tilghman (license 14) + 20100811__issue16770.diff.txt uploaded by tilghman (license 14) + Tested by: jamicque + +2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes no default transport for temp peer + creation in chan_sip (closes issue #17829) Reported by: falves11 + Patches: issue_17829.rev1.txt uploaded by russell (license 2) + issue_17829.diff uploaded by dvossel (license 671) Tested by: + falves11 + +2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen@digium.com> + + * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information + about creating sounds files using the sounds tools publically + available so that others can create their own sounds prompts + using the same tools we use to generate sounds releases. This + allows people creating their own prompts to sound consistent with + the prompts available from the open source project. SWP-595 + +2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson@digium.com> + + * main/channel.c, /: Merged revisions 282430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) + | 16 lines Send a SRCCHANGE indication when we masquerade + Masquerading a channel means that the src of the audio is + potentially changing, so send a SRCCHANGE so that RTP-based media + streams can get a new SSRC generated to reflect the change. + Original patch by addix (along with lots of testing--thanks!). + (closes issue #17007) Reported by: addix Patches: + 1001-reset-SSRC-original-channel.diff uploaded by addix (license + 1006) srcchange.diff uploaded by twilson (license 396) Tested by: + addix, twilson Review: https://reviewboard.asterisk.org/r/862/ + ........ + +2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: only do magic pickup when notifycid is + enabled A new way of doing BLF pickup was introduced into 1.6.2. + This feature adds a call-id value into the XML of a SIP_NOTIFY + message sent to alert a subscriber that a device is ringing. This + option should only be enabled when the new 'notifycid' option is + set... but this was not the case. Instead the call-id value was + included for every RINGING Notify message, which caused a + regression for people who used other methods for call pickup. + (closes issue #17633) Reported by: urosh Patches: chan_sip.txt + uploaded by urosh (license ) blf_cid_issue.diff uploaded by + dvossel (license 671) Tested by: dvossel, urosh, okrief, + alecdavis + +2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker@digium.com> + + * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | + 1 line Register CLI commands before parsing config, in case there + is a config error. ........ + +2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 281911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) + | 20 lines Ensure SSRC is changed when media source is changed to + resolve audio delay. This change causes the SSRC to change right + before the channels are bridged, which is what used to happen. It + seems that fixes were made to attempt limiting SSRC changes, + targeted mainly at sending DTMF. DTMF is not affecting the SSRC + with this change. There are two other control frames sent in + ast_channel_bridge that probably should also be changed to + AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change + up to the discretion of resolving issue #17007. For reference - + old review implementing new control frame SRCCHANGE: + https://reviewboard.asterisk.org/r/540 (closes issue #17404) + Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler + (license 325) Tested by: sdolloff ........ + +2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen@digium.com> + + * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 + Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes + issue #17836) Reported by: RoadKill Patches: + say.conf.sample.patch.dk uploaded by RoadKill (license 933) + ........ + + * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 + Aug 2010) | 6 lines Allow say.conf to handle large numbers ending + with multiple zeros. (closes issue #17833) Reported by: RoadKill + Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill + (license 933) ........ + +2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher@digium.com> + + * apps/app_readexten.c: Only set status TIMEOUT, if we have no + digits. (closes issue #15188) Reported by: jcovert Patches: + app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license + 551) + +2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell@digium.com> + + * main/sched.c: Don't move the time threshold for running scheduled + events on every iteration. Instead, only calculate the time + threshold each time ast_sched_runq() is called. (closes issue + #17742) Reported by: schmidts Patches: sched.c.patch uploaded by + schmidts (license 1077) + + * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) + | 8 lines Reset visible indication after answer. (closes issue + #17641) Reported by: klaus3000 Patches: + ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by + klaus3000 (license 65) Tested by: schmidts ........ + +2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out + the peer's addr before it was removed from the peers_by_ip + container. This made it impossible to be removed from the + container as the addr is the key used by the container to find + the peer. (closes issue #17774) Reported by: kkm Patches: + 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) + 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) + +2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 281390 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 + Aug 2010) | 13 lines Prevent loss of Caller ID information set on + local channel after masquerade. Caller ID set on the channel + before a masquerade occurs when using a local channel would cause + the information to be lost. The problem was that the information + was set on a channel destined to be hung up. The somewhat + confusing fix is to detect if any Caller ID has been set on the + channel and if so preswap the Caller ID data so that basically + the masquerade puts the data back. (closes issue #17138) Reported + by: kobaz Review: https://reviewboard.asterisk.org/r/847/ + ........ + +2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell@digium.com> + + * main/cdr.c: Cleanup default option value handling for cdr.conf + [general]. The default values would differ depending on whether + or not cdr.conf exists. That is no longer the case. Apply a + default value to the unanswered option. Define all default values + as named constants. + +2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) + | 8 lines Change context lock back to a mutex, because + functionality depends upon the lock being recursive. (closes + issue #17643) Reported by: zerohalo Patches: + 20100726__issue17643.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ + +2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions + 280811 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) + | 9 lines Prevent DAHDI channels from overriding the callerid, + once it's been set by the user. (closes issue #16661) Reported + by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by + tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt + uploaded by tilghman (license 14) Tested by: jstapleton ........ + + * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document + -B and -W flags and regenerate manpage from sgml + + * apps/app_voicemail.c: Allow the pipe, but also allow the comma + +2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_sip.c: Change SIP NOTIFY requests to expect a + response so authentication will work. This changes the request to + be sent with the transmit type XMIT_RELIABLE so that sip_ack + doesn't return false and cause the 401 to be ignored in cases + where authentication is required. (closes issue #14255) Reported + by: zktech + +2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c: Off-by-one error (closes issue #17590) + Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded + by tilghman (license 14) + +2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes wrong SRV query for TLS connection + (closes issue #17612) Reported by: marcelloceschia Patches: + chan-sip_srvQuery.patch uploaded by marcelloceschia (license + 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) + chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia + (license 1079) Tested by: marcelloceschia, st, pabelanger + + * main/channel.c, /: Merged revisions 280448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) + | 12 lines fixes issue with translator frame not getting freed A + translator frame even if it local storage so the translation path + can be freed. This issue prevented g729 licenses from being freed + up. (closes issue #17630) Reported by: manvirr Patches: + encoder_fix.diff uploaded by dvossel (license 671) Tested by: + manvirr, dvossel ........ + +2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau@digium.com> + + * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | + 2 lines Fix a dsp structure leak occuring when a local channel is + put into a meetme conference, then masquaraded away. ABE-2422 + ........ + +2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_local.c: Implement support for + ast_channel_queryoption on local channels. Currently only + AST_OPTION_T38_STATE is supported. ABE-2229 Review: + https://reviewboard.asterisk.org/r/813/ + +2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker@digium.com> + + * sounds/Makefile: Work around some silly behavior on BSD. A + non-zero exit from a subshell should make the build fail. (closes + issue #17621) + +2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Add missing enum value "unknown" to the + SS7 called_nai and calling_nai config options. + +2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker@digium.com> + + * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is + no util on the system. + + * sounds/Makefile: Remove unnecessary subshells. Attempt to make + checksumming work. Also improves readability. (issue #17621) + Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ + +2010-07-28 16:51 +0000 [r280160] Sean Bright <sean@malleable.com> + + * apps/app_queue.c: Plug a reference leak in app_queue when adding + members dynamically. (closes issue #17738) Reported by: + bobwienholt Patches: issue17738.patch uploaded by bobwienholt + (license 950) Tested by: bobwienholt, seanbright + +2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 + Jul 2010) | 1 line Update help text to be less confusing. + ........ + +2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel@digium.com> + + * main/audiohook.c, main/channel.c, /, + include/asterisk/audiohook.h: Merged revisions 279945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) + | 19 lines remove empty audiohook write list on channel If a + channel has an audiohook write list created on it, that list + stays on the channel until the channel is destroyed. There is no + reason to keep that list on the channel if it becomes empty. If + it is empty that just means we are doing needless translating for + every ast_read and ast_write. This patch removes the audiohook + list from the channel once it is detected to be empty on either a + read or write. If a audiohook is added back to the channel after + this list is destroyed, the list just gets recreated as if it + never existed to begin with. (closes issue #17630) Reported by: + manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ + +2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker@digium.com> + + * makeopts.in, configure, configure.ac: Add SHA1SUM to configure, + since we require it for sounds/ + + * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they + aren't used anymore. + + * sounds/Makefile: Simply sounds/Makefile some more. + +2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static + option in sip.conf. We were attempting to create a contactdeny + rule based on the peer's IP address before the peer's IP address + had been set. By moving the processing further down in the + function, we can ensure stuff works as we expect for it to. + (closes issue #17717) Reported by: mmichelson Patches: + 17717.patch uploaded by mmichelson (license 60) Tested by: + DennisD + +2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker@digium.com> + + * sounds/Makefile (added), sounds/Makefile.380 (removed), + configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 + (removed), configure.ac: Really fix sounds Makefile (and make it + readableish). There was a rather large syntax error that should + have caused ALL versions of GNU make to fail. I don't know how it + worked. + +2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher@digium.com> + + * configure, configure.ac: Dunno why this worked on my machine, but + it works better this way. + +2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry@suretecsystems.com> + + * res/res_config_ldap.c: Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 (closes issue + #13573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec + +2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher@digium.com> + + * sounds/Makefile (removed), configure, sounds/Makefile.380 + (added), sounds/Makefile.381 (added), configure.ac: Use a special + Makefile for noobs who still have GNU Make 3.80. (Closes issue + #17716) Reported by: farisraouf + +2010-07-26 15:41 +0000 [r279501] Sean Bright <sean@malleable.com> + + * autoconf/ast_ext_lib.m4: Expand the correct value within + AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth + +2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus@gmail.com> + + * doc/asterisk.8: Minor update to man page + +2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett@digium.com> + + * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) + | 7 lines SIP promiscuous redirect could fail to dial the + redirect. The ast_channel was created with one variable to + ast_request() but the call to ast_call() that initiates the + outgoing call was using a different variable. The two variables + are not equivalent if the call_forward string included a channel + technology specifier. e.g., SIP/200 ........ + +2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk + to 1.6.2. + +2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell@digium.com> + + * /: remove old properties + + * /: Add branch-1.4-merged and branch-1.4-blocked properties to + 1.6.2 branch. + +2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher@digium.com> + + * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged + revisions 278985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010) + | 12 lines Merged revisions 278984 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) + | 5 lines Establish a maximum version for openh323 (i.e. not + opal), because chan_h323 will fail to load, even if it links. + (issue #17679) Reported by: am ........ ................ + + * main/asterisk.c, /: Merged revisions 278982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010) + | 15 lines Merged revisions 278981 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) + | 8 lines Avoid race with consolethread on shutdown (on parallel + processors). (closes issue #17080) Reported by: sybasesql + Patches: 20100721__issue17080.diff.txt uploaded by tilghman + (license 14) Tested by: sybasesql ........ ................ + +2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_dahdi.c: Two more typos to cancell. + +2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler@digium.com> + + * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 | + jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines + Add method for finding XML doc files for systems that don't + support GLOB_BRACE. In particular, Solaris and perhaps others do + not support the above mentioned GNU extension. In this case the + paths are simply expanded without the braces and the calls to + glob are made separately. Note: I could not explain memory + allocation failures that were being reported from within libxml + itself when making calls to glob without using GLOB_NOCHECK. This + is the only reason why that flag is being used. (closes issue + #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded + by snuffy (license 35), modified by me ........ + +2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: DNID does not get cleard on a new call + when using immediate=yes with ISDN signaling. When you are using + chan_dahdi ISDN signaling with immediate=yes and a call comes in + without a DNID then you get the DNID of a previous call. + Chan_dahdi does not touch the DNID field on a new call if it does + not have a DNID. Made always copy the DNID from the new call. The + patches backport the relevant changes from trunk -r210387. + (closes issue #17568) Reported by: wuwu Patches: + issue17568_v1.4.patch uploaded by rmudgett (license 664) + issue17568_v1.6.2.patch uploaded by rmudgett (license 664) + +2010-08-10 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.11 Released. + +2010-07-26 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.11-rc2 Released. + +2010-07-26 Leif Madsen <lmadsen@digium.com> + + * qwell, asterisk, branch-1.6.2, r279657 *** + Really fix sounds Makefile (and make it readableish). + There was a rather large syntax error that should have + caused ALL versions of GNU make to fail. + I don't know how it worked. + + (Closes issue #17716) + +2010-07-22 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.11-rc1 Released. + +2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 278620 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul + 2010) | 19 lines Merged revisions 278618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul + 2010) | 13 lines Allow PLC to function properly when channels use + SLIN for audio. If a channel involved in a bridge was using SLIN + audio, then translation paths were not guaranteed to be set up + properly since in all likelihood the number of translation steps + was only 1. This patch enforces the transcode_via_slin behavior + if transcode_via_slin or generic_plc is enabled and one of the + formats to make compatible is SLIN. AST-352 ........ + ................ + +2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO + channels This fixes some cases of no outgoing calls on FXO before + an incoming call. Remove an unnecessary testing of an "off-hook" + bit from DAHDI for FXO (KS/GS) channels.In some cases the bit + would not be initialized properly before the first inbound call + and thus prevent an outgoing call. If those tests are actually + required by anybody, they should define DAHDI_CHECK_HOOKSTATE in + channels/sig_analog.c . (closes issue #14577) Reported by: jkroon + Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd + (license 610) Tested by: frawd Review: + https://reviewboard.asterisk.org/r/699/ + +2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell@digium.com> + + * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) + | 41 lines Use poll() instead of select() in res_timing_pthread + to avoid stack corruption. This code did not properly check + FD_SETSIZE to ensure that it did not try to select() on fds that + were too large. Switching to poll() removes the limitation on the + maximum fd value. (closes issue #15915) Reported by: keiron + (closes issue #17187) Reported by: Eddie Edwards (closes issue + #16494) Reported by: Hubguru (closes issue #15731) Reported by: + flop (closes issue #12917) Reported by: falves11 (closes issue + #14920) Reported by: vrban (closes issue #17199) Reported by: + aleksey2000 (closes issue #15406) Reported by: kowalma (closes + issue #17438) Reported by: dcabot (closes issue #17325) Reported + by: glwgoes (closes issue #17118) Reported by: erikje possibly + other issues, too ... ........ + +2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 | + tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 + lines Ensure realtime conferences are treated the same as static + conferences when trying to find an empty one. Also, parse the + useropts properly, when retrieving from realtime, and add them to + the existing flags. (closes issue #17502) Reported by: kenji + Patches: 20100720__issue17502.diff.txt uploaded by tilghman + (license 14) Tested by: kenji ........ + + * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500 + (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) + | 7 lines Delete IMAP messages in reverse order, to ensure + reordering after each expunge does not cause deletion of the + wrong message. (closes issue #16350) Reported by: noahisaac + Patches: 20100623__issue16350.diff.txt uploaded by tilghman + (license 14) ........ ................ + + * main/autoservice.c, /, main/features.c, + include/asterisk/channel.h: Merged revisions 278272 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500 + (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) + | 4 lines Do not queue up DTMF frames while a call is on hold. + (Fixes ABE-2110) ........ ................ + + * main/manager.c, /: Merged revisions 278024 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) + | 14 lines Merged revisions 278023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) + | 7 lines Off-by-one error (closes issue #16506) Reported by: + nik600 Patches: 20100629__issue16506.diff.txt uploaded by + tilghman (license 14) ........ ................ + +2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau@digium.com> + + * /, main/features.c: Merged revisions 277945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | + 15 lines Merged revisions 277906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | + 7 lines Avoid trying to pickup a parked extension before the park + operation is completed. A crash could occur if the extension is + picked up while the parking extension is being announced. Testing + pu->notquiteyet while searching for a parked extension resolves + this crash. (ABE-2418) ........ ................ + +2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_pgsql.c: Merge issues... + + * /, autoconf/ast_func_fork.m4, configure, + include/asterisk/autoconfig.h.in: Merged revisions 277775 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500 + (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) + | 5 lines Remove uclibc cross-compile triplet, as uclibc has a + working fork()... it's only uclinux that does not. (closes issue + #17616) Reported by: pprindeville ........ ................ + + * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged + revisions 277773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010) + | 15 lines Merged revisions 277568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) + | 8 lines Since we split values at the semicolon, we should store + values with a semicolon as an encoded value. (closes issue + #17369) Reported by: gkservice Patches: + 20100625__issue17369.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + +2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach@gmail.com> + + * /, main/features.c: Merged revisions 277657 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul + 2010) | 16 lines Merged revisions 277625 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul + 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on + attended transfer. ast_bridge_call() clears + AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, + ast_bridge_call() is called for a second bridge on the same + channel, and it clears that flag, which still needs to get set + for when the original ast_bridge_call() gets control back and + checks it. Review: https://reviewboard.asterisk.org/r/741 + ........ ................ + +2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul + 2010) | 11 lines Merged revisions 277497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul + 2010) | 4 lines Default to no udptl error correction so that + error correction will be disabled in the event that the remote + end indicates that they do not support the error correction mode + we requested. FAX-128 ........ ................ + +2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler@digium.com> + + * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 | + jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines + Fix reporting estimated queue hold time. Just say the number of + seconds (after minutes) rather than doing some incorrect + calculation with respect to minutes. (closes issue #17498) + Reported by: corruptor Patches: holdesecs_bug.diff uploaded by + corruptor (license 253) ........ + +2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500 + (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) + | 15 lines priexclusive in chan_dahdi.conf ignored when reloading + dahdi module During a reload, the priexclusive and outsignalling + parameters are not read in from the config file as intended. + Unfortunately, they get set to defaults as a result. This patch + makes sure that they do not get set to defaults during a reload. + (closes issue #17441) Reported by: mtryfoss Patches: + issue17441_v1.4.patch uploaded by rmudgett (license 664) + issue17441_v1.6.2.patch uploaded by rmudgett (license 664) + issue17441_trunk.patch uploaded by rmudgett (license 664) Tested + by: rmudgett ........ ................ + +2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher@digium.com> + + * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql + (added), /: Merged revisions 277452 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 | + tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines + Add documentation for MOH realtime fields ........ + +2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler@digium.com> + + * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 | + jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines + Add missing handling for ringing state for use with queue empty + options. (closes issue #17471) Reported by: jazzy Patches: + app_queue.c.diff uploaded by jazzy (license 1056) ........ + +2010-07-16 18:33 +0000 [r277338] Matthew Nicholson <mnicholson@digium.com> + + * main/pbx.c, /: Merged revisions 277331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul + 2010) | 15 lines Merged revisions 277327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul + 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as + extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) + Reported by: francesco_r Patches: pbx.c.patch uploaded by + viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar + ........ ................ + +2010-07-16 18:14 +0000 [r277264] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 277263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) + | 12 lines Merged revisions 277261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) + | 5 lines If variable gotten is not set, will segfault on + Solaris. (closes issue #17636) Reported by: bklang ........ + ................ + +2010-07-16 17:31 +0000 [r277256] Matthew Nicholson <mnicholson@digium.com> + + * main/channel.c, /: Merged revisions 277250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul + 2010) | 11 lines Merged revisions 277247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul + 2010) | 4 lines For pass through DTMF tones, measure the actual + duration between the begin and end packets on the wire. If it is + detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf + emulation. AST-362 ........ ................ + +2010-07-16 17:18 +0000 [r277188] Paul Belanger <paul.belanger@polybeacon.com> + + * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul + 2010) | 15 lines Merged revisions 277182 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul + 2010) | 8 lines Total analysis time error with SIP and silence + suppression When using app_amd with SIP providers that have + silence suppression on, the iTotalTime count increases + exponentially. (closes issue #17656) Reported by: juls ........ + ................ + +2010-07-16 15:21 +0000 [r277144] Sean Bright <sean@malleable.com> + + * /, main/translate.c: Merged revisions 277143 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 | + seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8 + lines Avoid crashing when installing a duplicate translation path + with a lower cost. (closes issue #17092) Reported by: moy + Patches: translate.rev254273.patch uploaded by moy (license 222) + Tested by: moy ........ + +2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 | + jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines + Correct not setting the bindport before attempting to open the + socket. Related to changes from 276571, I was accidentally + testing with a port set in my configuration causing me to miss + this. Also moved the TCP handling as well to occur before + build_peer is called. ........ + + * main/channel.c, /: Merged revisions 276653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010) + | 9 lines Merged revisions 276652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) + | 2 lines In a perfect world, the frame source would never be + NULL. In the meantime, don't crash when it is. ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 | + jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines + Fix MWI notification transmission problems over SIP. MWI updates + were not being sent if no messages were found in the event cache. + This was corrected since a phone may need to clear its MWI status + configured previously from another mailbox. Upon module or sip + reload, MWI updates could not be sent due to the sipsock socket + not being set early enough in reload_config. The code handling + the descriptor assignment and such has simply been moved before + the call to build_peer. Issuing a sip reload cleared the IP + address of the peer, but skipped checking the database for + registration information. The database is now checked both for + sip reload and actually reloading the module. If a transmission + occurs before the do_monitor thread has started, do not attempt + to send a signal to it. (closes issue #17398) Reported by: ip-rob + ........ + +2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming <kpfleming@digium.com> + + * main/loader.c, /: Merged revisions 276441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 | + kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 + lines Don't try to call an embedded module's backup_globals() + function until after confirming it exists. ........ + +2010-07-14 11:52 +0000 [r276269] Leif Madsen <lmadsen@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 276268 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500 + (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 + Jul 2010) | 1 line Update documentation for voicemail.conf + externpass option. ........ ................ + +2010-07-13 19:11 +0000 [r276125] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 276124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010) + | 9 lines Merged revisions 276123 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) + | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr + instead of peer_cdr in the last commit). ........ + ................ + +2010-07-13 19:01 +0000 [r276121] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) + | 19 lines Merged revisions 275773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) + | 12 lines Make user removals and traversals thread safe in + meetme. Race conditions present in meetme involving the user list + where a lack of locking has the potential for a user to be + removed during a traversal or as in the case of the reporter + after checking if the list is empty could cause a crash. Fixing + this was done by convering the userlist to an ao2 container. + (closes issue #17390) Reported by: Vince Review: + https://reviewboard.asterisk.org/r/746/ ........ ................ + +2010-07-13 16:55 +0000 [r275996] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 275995 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010) + | 21 lines Merged revisions 275994 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) + | 14 lines Access peer->cdr directly instead of through a saved + off reference. At this point in the code, it is possible that + peer_cdr may be invalid. Specifically, in the blind transfer + code, CDRs are swapped between channels. So, peer_cdr is no + longer == peer->cdr. The scenario that exposed a crash in this + code was a blind transfer that hit the system call limit, causing + the transferee channel to get destroyed after the transfer + attempt failed. Even if it succeeds and this code doesn't crash, + this code was still trying to reset a CDR on a channel that was + now owned by a different thread, which is a BadThing(tm). + (ABE-2417) ........ ................ + +2010-07-13 14:49 +0000 [r275911] Tilghman Lesher <tlesher@digium.com> + + * contrib/realtime/mysql, contrib/realtime/oracle, + contrib/scripts/sip-friends.sql (removed), + contrib/realtime/mysql/sipfriends.sql, + contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql + (removed), contrib/realtime/mysql/meetme.sql, + contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql + (removed), contrib/scripts/iax-friends.sql (removed), /, + contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql + (removed), contrib/realtime (added), contrib/realtime/postgresql, + contrib/realtime/postgresql/realtime.sql: Merged revisions 275910 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500 + (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 + Jul 2010) | 2 lines Move SQL scripts into their own + database-specific directories. ........ ................ + +2010-07-12 17:26 +0000 [r275706] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 275682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010) + | 18 lines Merged revisions 275665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) + | 11 lines Change ast_write to not stop generator when called + from ast_prod. For SIP channels configured with the + progressinband option on, the ringback was being immediately + stopped. This problem was due to ast_prod being moved for a + deadlock fix in 259858. Prodding the channel after setting up the + generator triggered the check in ast_write to stop the generator. + The fix here should write the frame the same as was done before + the call to ast_prod was moved. (closes issue #17372) Reported + by: tech_admin ........ ................ + +2010-07-12 15:38 +0000 [r275627] Leif Madsen <lmadsen@digium.com> + + * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 | + lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines + cdr_pgsql does not detect when a table is found. This change adds + an ERROR message to let you know when a failure exists to get the + columns from the pgsql database, which typically means that the + table does not exist. (closes issue #17478) Reported by: kobaz + Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested + by: kobaz, russell, lmadsen ........ + +2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant <russell@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 245192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 | + mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 + lines Remove useless sip options related to hash table size. + First off, these options weren't actually doing anything. By the + time the options were parsed, the peer and dialog containers had + already been allocated with their default values. Second, hash + table size is something that doesn't really make sense to change + in a config file. If a user is that interested in changing the + hashtable size, he can modify the source itself. I have removed + the parsing of the hash_peer, hash_user, and hash_dialog options. + I have removed the hash_user_size variable altogether since it is + not used at all. I also changed hash_peer_size and + hash_dialog_size to be constant, and have changed the symbols to + be in all caps as constants typically are. I have also removed + the entire section in sip.conf.sample regarding configurable + hashtable sizes. ........ (merge to 1.6.2 inspired by issue + #17553) + + * /: unblock a rev + + * configs/features.conf.sample, /, main/features.c: Merged + revisions 275424 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 | + russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines + Fix some issues related to dynamic feature groups in + features.conf. The bridge handling code did not properly consider + feature groups when setting parameters that would affect whether + or not a native bridge would be attempted. If DYNAMIC_FEATURES + only include a feature group, a native bridge would occur that + may prevent features from working. Fix a bug in verbose output + that would show the key mapping as empty if it was using the + default mapping and not a custom mapping in the feature group. + Add feature groups to the output of "features show". Adjust the + feature execution logic to match that of the logic when executing + a feature that was not configured through a feature group. Update + features.conf.sample to show that an '=' is still required if + using the default key mapping from [applicationmap]. Finally, + clean up a little bit of formatting to better coform to coding + guidelines while in the area. (closes issue #17589) Reported by: + lmadsen Patches: issue_17589.rev4.txt uploaded by russell + (license 2) Tested by: russell, lmadsen ........ + + * /, main/features.c: Merged revisions 275310 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 | + russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines + Add missing ao2_iterator_destroy(). ........ + +2010-07-09 19:23 +0000 [r275260] Paul Belanger <paul.belanger@polybeacon.com> + + * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul + 2010) | 15 lines Merged revisions 275241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul + 2010) | 8 lines Fix logging message for stale nonce. (closes + issue #17582) Reported by: kenner Patches: chan_sip.c.diff + uploaded by kenner (license 1040) Tested by: lmadsen ........ + ................ + +2010-07-09 18:24 +0000 [r275191] Matthew Nicholson <mnicholson@digium.com> + + * main/loader.c, /: Merged revisions 275186 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul + 2010) | 9 lines Merged revisions 275182 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul + 2010) | 2 lines give a better error message when attempting to + unload a module that is not loaded ........ ................ + +2010-07-09 18:11 +0000 [r275148] Russell Bryant <russell@digium.com> + + * configs/features.conf.sample, /: Merged revisions 275147 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09 + Jul 2010) | 2 lines Move parking lot sample config out from the + middle of dynamic features sample config. ........ + +2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson <mnicholson@digium.com> + + * main/loader.c, /: Merged revisions 275144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul + 2010) | 9 lines Merged revisions 275143 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul + 2010) | 2 lines don't unload modules that returned + AST_MODULE_LOAD_DECLINE when they were loaded ........ + ................ + + * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul + 2010) | 15 lines Merged revisions 275027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul + 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels + going into the pbx via the G option in app_dial (closes issue + #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff + uploaded by mnicholson (license 96) Tested by: jamicque, + mnicholson ........ ................ + +2010-07-09 15:39 +0000 [r275023] Russell Bryant <russell@digium.com> + + * include/asterisk/test.h, /, main/test.c: Merged revisions 275022 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r275022 | russell | 2010-07-09 10:35:53 -0500 + (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) + | 4 lines Document that a leading and trailing slash is expected + for test categories. Also, emit a warning if a test is registered + without one of these. ........ ................ + +2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010) + | 1 line Add missing conditional around chan_dahdi + mfcr2_skip_category config parameter. ........ + + * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500 + (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 + Jul 2010) | 1 line Close the DAHDI FD on error when processing + chan_dahdi toneduration config parameter. ........ + ................ + +2010-07-07 06:16 +0000 [r274419] Tilghman Lesher <tlesher@digium.com> + + * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500 + (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) + | 8 lines Correct how 100, 200, 300, etc. is said. Also add the + crazy British numbers. (closes issue #16102) Reported by: Delvar + Patches: say.conf.fix.patch uploaded by Delvar (license 908) + (plus a few additional fixes and simplifications by me) ........ + ................ + +2010-07-06 23:06 +0000 [r274360] Terry Wilson <twilson@digium.com> + + * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions + 274284 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) + | 18 lines Merged revisions 274280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) + | 9 lines Add option to not do a call forward on 482 Loop + Detected Asterisk has always set up a forwarded call when + receiving a 482 Loop Detected. This prevents handling the call + failure by just continuing on in the dialplan. Since this would + be a change in behavior, the new option to disable this behavior + is forwardloopdetected which defaults to 'yes'. Review: + https://reviewboard.asterisk.org/r/764/ ........ ................ + +2010-07-06 22:30 +0000 [r274347] Jeff Peeler <jpeeler@digium.com> + + * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500 + (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) + | 7 lines Correct sip.conf.sample comments for prematuremedia + option. (closes issue #17513) Reported by: festr Patches: patch + uploaded by festr (license 443) ........ ................ + +2010-07-06 22:10 +0000 [r274282] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010) + | 2 lines Status shows all non-CRC4 lines as "yellow", even if + "yellow" was not in the bitfield. ........ + +2010-07-06 14:33 +0000 [r274168] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Merged revisions 274164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul + 2010) | 22 lines Merged revisions 274157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul + 2010) | 16 lines Fix problem with RFC 2833 DTMF not being + accepted. A recent check was added to ensure that we did not + erroneously detect duplicate DTMF when we received packets out of + order. The problem was that the check did not account for the + fact that the seqno of an RTP stream will roll over back to 0 + after hitting 65535. Now, we have a secondary check that will + ensure that the seqno rolling over will not cause us to stop + accepting DTMF. (closes issue #17571) Reported by: mdeneen + Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license + 60) Tested by: richardf, maxochoa, JJCinAZ ........ + ................ + +2010-07-05 13:55 +0000 [r273888] Paul Belanger <paul.belanger@polybeacon.com> + + * main/config.c, /: Merged revisions 273886 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul + 2010) | 15 lines Merged revisions 273884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul + 2010) | 8 lines Remove extra line breaks from 'core show config + mappings' (closes issue #17583) Reported by: pabelanger Patches: + issue17583.patch uploaded by pabelanger (license 224) Tested by: + lmadsen ........ ................ + +2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /, channels/chan_agent.c, + channels/chan_h323.c, include/asterisk/lock.h: Merged revisions + 273830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) + | 16 lines Merged revisions 273793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) + | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock + fails, to help catch potentially large software bugs. (closes + issue #17407) Reported by: pdf Patches: + 20100527__issue17407.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/751/ ........ + ................ + + * main/autoservice.c, /: Merged revisions 273718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010) + | 15 lines Merged revisions 273717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) + | 8 lines Autoservice loop optimization causes a busy loop, when + channels are serviced while in hangup. (closes issue #17564) + Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt + uploaded by tilghman (license 14) Tested by: ramonpeek ........ + ................ + + * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 | + tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines + The switch fallthrough could create some errorneous situations, + so best to force directly to the default case. ........ + +2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_iax2.c, apps/app_voicemail.c, + channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix + typos reported by Lintian + +2010-07-01 22:17 +0000 [r273571] Russell Bryant <russell@digium.com> + + * main/datastore.c, /: Merged revisions 273566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010) + | 14 lines Merged revisions 273565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) + | 7 lines Don't return a partially initialized datastore. If + memory allocation fails in ast_strdup(), don't return a partially + initialized datastore. Bad things may happen. (related to + ABE-2415) ........ ................ + +2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010) + | 21 lines Merged revisions 273474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) + | 14 lines Allow admin user to join conference without using + admin mode and no user pin. Configuring the conference in + meetme.conf like the following: conf => 2345,,6666 did not prompt + for pin when used without admin mode. This meant that the + conference could not be joined as an admin even if the user knew + the correct pin. The original bug report was submitted claiming + that the blank user pin should deny entry into the conference. I + think a better way to handle this would be with a feature + enhancement that used the following syntax: conf => 2345,X,6666 - + where X denotes no acceptable pin allowed (closes issue #15704) + Reported by: modelnine ........ ................ + + * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010) + | 19 lines Merged revisions 273354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) + | 12 lines Ensure channel placed in meetme in ringing state is + properly hung up. An outgoing channel placed in meetme while + still ringing which was then hung up would not exit meetme and + the channel was not properly destroyed. Specifically checking for + this scenario by looking at the appropriate control frames + resolves the issue. (closes issue #15871) Reported by: Ivan + Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan + (license 229) ........ ................ + +2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson <mnicholson@digium.com> + + * main/manager.c, /: Merged revisions 273352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 | + mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2 + lines Fixed whitespace problems ........ + + * main/manager.c, /: Merged revisions 273350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 | + mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2 + lines Altered my comment about TCP_NODELAY ........ + + * main/manager.c, /: Merged revisions 273270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 | + mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2 + lines Set TCP_NODELAY on manager TCP sockets to prevent delays on + outgoing packets. This regression was introduced in r48338. + AST-359 ........ + +2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger <paul.belanger@polybeacon.com> + + * main/rtp.c, /: Merged revisions 273233 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 | + pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11 + lines Fix rt(c)p set debug ip taking wrong argument Also clean up + some coding errors. (closes issue #17469) Reported by: wdoekes + Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes + (license 717) Tested by: wdoekes, pabelanger ........ + + * /: Revert previous commit; res_rtp_asterisk.c does not exist. + + * /: Unblock revisions 218107 ........ r218107 | mvanbaak | + 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the + actual given ip address for 'rtp set debug ip <foo>' instead of + the word 'ip' (closes issue 0015711) Reported by: davidw Patches: + 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) + Tested by: davidw ........ + +2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 273144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 | + tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines + Permission checking for the system application is backwards. + (closes issue #17550) Reported by: kenner Patches: manager.c.diff + uploaded by kenner (license 1040) Tested by: kenner ........ + + * main/config.c, /: Merged revisions 273142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 | + tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines + Don't attempt to proceed if our internal parser indicates an + invalid file. (closes issue #17560) Reported by: Nick_Lewis + ........ + + * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010) + | 17 lines Merged revisions 273060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) + | 10 lines Allow the "useragent" value to be restored into memory + from the realtime backend. This value is purely informational. It + does not alter configuration at all. (closes issue #16029) + Reported by: Guggemand Patches: realtime-useragent.patch uploaded + by Guggemand (license 897) Tested by: Guggemand ........ + ................ + + * main/channel.c, /: Merged revisions 273058 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010) + | 11 lines Recorded merge of revisions 273057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) + | 4 lines _Really_ skip the channel... don't just retry for + another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........ + ................ + + * main/pbx.c, /: Merged revisions 273054 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 | + tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11 + lines Send DialPlanComplete as a response, not as a separate + event. Otherwise, it goes to all manager sessions and may exclude + the current session, if the Events mask excludes it. (closes + issue #17504) Reported by: rrb3942 Patches: + showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested + by: rrb3942 ........ + +2010-06-29 16:43 +0000 [r272972] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 253357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 | + russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines + Increase CLI command output timeout for asterisk -rx to 60 + seconds. (closes issue #17049) Reported by: russell Tested by: + russell Review: https://reviewboard.asterisk.org/r/573/ ........ + +2010-07-22 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.10 + + * Included a fix for res_timing_pthread per the description below: + + r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines + + Use poll() instead of select() in res_timing_pthread to avoid stack corruption. + This code did not properly check FD_SETSIZE to ensure that it did not try to + select() on fds that were too large. Switching to poll() removes the limitation + on the maximum fd value. + +2010-07-07 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.10-rc2 + + * Fix problem with RFC 2833 DTMF not being accepted. + + A recent check was added to ensure that we did not erroneously + detect duplicate DTMF when we received packets out of order. + The problem was that the check did not account for the fact that + the seqno of an RTP stream will roll over back to 0 after hitting + 65535. Now, we have a secondary check that will ensure that the + seqno rolling over will not cause us to stop accepting DTMF. + + (closes issue 0017571) + Reported by: mdeneen + Patches: + rtp_seqno_rollover.patch uploaded by mmichelson (license 60) + Tested by: richardf, maxochoa, JJCinAZ + + * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx + via the G option in app_dial + + (closes issue 0017592) + Reported by: jamicque + Patches: + G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) + Tested by: jamicque, mnicholson + +2010-06-29 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.10-rc1 + +2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 272926 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010) + | 15 lines Merged revisions 272925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) + | 8 lines Don't change ownership/group/permissions on run + directory, if it already exists. (closes issue #17076) Reported + by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by + tilghman (license 14) Tested by: stuarth ........ + ................ + + * main/config.c, /: Merged revisions 272923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010) + | 19 lines Merged revisions 272921-272922 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) + | 8 lines Change the way that we read include files, to + accommodate for changes in GCC 4.4. (closes issue #17472) + Reported by: seandarcy Patches: config2.patch uploaded by nivan + (license 1066) Tested by: nivan ........ r272922 | tilghman | + 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim + trailing blanks on #includes ........ ................ + +2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell@digium.com> + + * tests/test_astobj2.c (added): Backport applicable parts of + test_astobj2 from trunk. + +2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun + 2010) | 11 lines Merged revisions 272804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun + 2010) | 5 lines Decode URI in contact header of 302 response. + ABE-2352 ........ ................ + +2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell@digium.com> + + * doc/tex/chan-mobile.tex (removed): remove accidentally added + file. + + * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /, + doc/tex/chan-mobile.tex (added): Merged revisions 272684 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28 + Jun 2010) | 2 lines Use the underscore package so that + underscores do not need to be escaped. ........ + +2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher@digium.com> + + * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500 + (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) + | 5 lines Make the structure of the table specified before match + the queries and results. (closes issue #17557) Reported by: cmaj + ........ ................ + + * sounds/Makefile, /: Merged revisions 272533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 | + tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines + Symlink sounds files, to save disk space, when multiple + tarballs/checkouts are on the same system. ........ + +2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell@digium.com> + + * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c, + include/asterisk/test.h (added), build_tools/cflags-devmode.xml, + tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c + (added): Backport unit test API from trunk. Also, update existing + test modules that were already in this branch but had been + converted to the unit test API in trunk. Review: + https://reviewboard.asterisk.org/r/748/ + +2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500 + (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) + | 10 lines ss_thread calls pri_grab without lock during overlap + dial Recent changes to chan_dahdi with relation to overlap + dialing call pri_grab without first obtaining a lock. (closes + issue #17414) Reported by: pdf Patches: bug17414.patch uploaded + by jpeeler (license 325) ........ ................ + +2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson@digium.com> + + * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions + 272254,272256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 | + twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines + Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes + issue #16991) Reported by: pprindeville Patches: + with_netsnmp.patch.txt uploaded by twilson (license 396) Tested + by: twilson Review: https://reviewboard.asterisk.org/r/739/ + ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23 + Jun 2010) | 2 lines Update configure when changing autconf m4 + files... ........ + +2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010) + | 23 lines Resolve some errors produced during module unload of + chan_iax2. The external test suite stops Asterisk using the "core + stop gracefully" command. The logs from the tests show that there + are a number of problems with Asterisk trying to cleanly shut + down. This patch addresses the following type of error that comes + from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 + __ast_pthread_mutex_destroy: chan_iax2.c line 11371 + (iax2_process_thread_cleanup): Error destroying mutex + &thread->lock: Device or resource busy For an example in the + context of a build, see: + http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary + purpose of this patch is to change the thread pool shutdown + procedure to be more explicit to ensure that the thread exits + from a point where it is not holding a lock. While testing that, + I encountered various crashes due to the order of operations in + unload_module() being problematic. I reordered some things there, + as well. Review: https://reviewboard.asterisk.org/r/736/ ........ + +2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun + 2010) | 16 lines Merged revisions 272367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 This version + of the patch only adds AgentComplete for attended transfers. It + was already present for blind transfers. ........ r272367 | + mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 + lines Send AgentComplete manager events in the event of blind and + attended transfers. (closes issue #16819) Reported by: elbriga + Patches: app_queue.diff uploaded by elbriga (license 482) + ........ ................ + +2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010) + | 8 lines If there is realtime configuration, it does not get + re-read on reload unless the config file also changes. (closes + issue #16982) Reported by: dmitri Patches: res_musiconhold.patch + uploaded by dmitri (license 1001) Tested by: atis ........ + +2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger@polybeacon.com> + + * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does + not exist in 1.6.2 + +2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher@digium.com> + + * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 | + tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines + Ensure a NULL file while debugging cannot crash AEL. (closes + issue #17215) Reported by: vazir Patches: + 20100518__issue17215.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ + +2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger@polybeacon.com> + + * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 | + pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2 + lines Fix previous merge. ast_test_flag != ast_test_flag64 + ........ + + * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun + 2010) | 19 lines Merged revisions 272255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun + 2010) | 12 lines First caller into a dynamic conference now enter + pin once. If MeetMe is configured to use dynamic conference + numbers, then the first caller (which creates the conference) had + to enter the PIN number twice. (closes issue #15878) Reported by: + shawkris Patches: issue15878.patch uploaded by pabelanger + (license 224) Tested by: pabelanger ........ ................ + + * main/manager.c, /: Merged revisions 272252 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 | + pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8 + lines Correct manager variable 'EventList' case. (closes issue + #17520) Reported by: kobaz Patches: manager.patch uploaded by + kobaz (license 834) Tested by: lmadsen ........ + +2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson@digium.com> + + * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 | + twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines + Don't start the sla thread unless we realy need it ........ + + * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 | + twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines + Make sure reload updates SLA config Even if there are no stations + or trunks defined, we need to start the sla thread to make sure + we get the reload event. Also, when doing a reload we need to + remove the existing trunks and stations or they end up hanging + around. (closes issue #16818) Reported by: mbonin Patches: + sla_reload.patch uploaded by twilson (license 396) Tested by: + twilson ........ + +2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel@digium.com> + + * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 | + dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines + fixes issue with 'dialplan remove extension blah' segfaulting + with tab completion (closes issue #17440) Reported by: kobaz + ........ + +2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun + 2010) | 15 lines Merged revisions 271902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun + 2010) | 8 lines Decrease the module ref count in sip_hangup when + SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the + ref count correct. (closes issue #16815) Reported by: rain + Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) + (modified) Tested by: rain ........ ................ + +2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell@digium.com> + + * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22 + Jun 2010) | 7 lines Resolve some errors that occur on a graceful + shutdown. Don't Finalize() if Initialize() did not succeed. This + resulted in an error about trying to Finalize() an invalid + handle. Also trim some trailing whitespace while in the area. + ........ + +2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel@digium.com> + + * /, main/features.c: Merged revisions 271831 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 | + dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines + fixes attended transfer behavior when both transferee and + transferer hung up If both the transferer and transferee of a + attended transfer hangup before the new channel picks up, the new + channel should be hung up as well as it has no endpoint to talk + to. This mirrors the expected behavior used in 1.4. (closes issue + #17444) Reported by: corruptor ........ + +2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson@digium.com> + + * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions + 271762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun + 2010) | 15 lines Merged revisions 271761 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun + 2010) | 9 lines Allow users to specify a port for dundi peers. + (closes issue #17056) Reported by: klaus3000 Patches: + dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) + Tested by: klaus3000 ........ ................ + + * include/asterisk/strings.h, configs/sip_notify.conf.sample, /, + channels/chan_sip.c: Merged revisions 271690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun + 2010) | 18 lines Merged revisions 271689 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun + 2010) | 8 lines Modify chan_sip's packet generation api to + automatically calculate the Content-Length. This is done by + storing packet content in a buffer until it is actually time to + send the packet, at which time the size of the packet is + calculated. This change was made to ensure that the + Content-Length is always correct. (closes issue #17326) Reported + by: kenner Tested by: mnicholson, kenner Review: + https://reviewboard.asterisk.org/r/693/ ........ This change also + adds an ast_str_copy_string() function (similar to + ast_copy_string), that copies one ast_str into another, properly + handling embedded nulls. ................ + +2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler@digium.com> + + * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010) + | 14 lines Merged revisions 271552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) + | 7 lines Do not use sizeof to calculate size of a heap allocated + character array. Change left out from 271399. (closes issue + #16053) Reported by: diLLec ........ ................ + +2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler@digium.com> + + * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged + revisions 271483 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010) + | 18 lines Merged revisions 271399 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) + | 11 lines Fix crash when parsing some heavily nested statements + in AEL on reload. Due to the recursion used when compiling AEL in + gen_prios, all the stack space was being consumed when parsing + some AEL that contained nesting 13 levels deep. Changing a few + large buffers to be heap allocated fixed the crash, although I + did not test how many more levels can now be safely used. (closes + issue #16053) Reported by: diLLec Tested by: jpeeler ........ + ................ + + * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010) + | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling + dahdi_indicate() within dahdi_fixup() while the owner pointers + are in a potentially inconsistent state is a potentially bad + thing in principle. However, calling dahdi_indicate() when the + channel private lock is already held can cause a deadlock if the + PRI lock is needed because dahdi_indicate() will also get the + channel private lock. The pri_grab() function assumes that the + channel private lock is held once to avoid deadlock. ........ + +2010-06-17 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.9 Released. + +2010-06-10 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.9-rc3 Released. + +2010-06-10 Tilghman Lesher <tlesher@digium.com> + + * Ensure signals are not blocked inside other signal handlers. + + This eliminates the annoying <beep> on the console. + + (closes issue 0017477) + Reported by: jvandal + Patches: + 20100610__issue17477.diff.txt uploaded by tilghman (license 14 + +2010-06-09 Paul Belanger <paul.belanger@polybeacon.com> + + * Fix Debian init script to not use -c. + + When using the init script as-is currently, it could cause issues on Debian + such as high CPU usage. This fix has worked for several people so I'm + implementing the change. We now handle color displays properly. + + (closes issue 0016784) + Reported by: pabelanger + Patches: + 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) + Tested by: pabelanger, tilghman + +2010-06-07 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.9-rc2 Released. + +2010-06-07 Tilghman Lesher <tlesher@digium.com> + + * Fix crash in DTMF detection. + + What I did not originally see in my previous commit was that even + though the next digit could be detected before the previous was + considered ended, the detection of the next digit effectively ends + the detection of the previous. Therefore, the length moves in + lockstep with the digit, and no separate counter is needed for the + length alone. + + (closes issue 0017371) + Reported by: alecdavis + + (closes issue 0017474) + Reported by: kenner + +2010-06-01 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.9-rc1 Released. + +2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 266592 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010) + | 18 lines Merged revisions 266585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) + | 11 lines Prevent CLI prompt from distorting output of lines + shorter than the prompt. Uses the VT100 method of clearing the + line from the cursor position to the end of the line: Esc-0K + (closes issue #17160) Reported by: coolmig Patches: + 20100531__issue17160.diff.txt uploaded by tilghman (license 14) + Tested by: coolmig ........ ................ + +2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com> + + * res/res_agi.c: Fix typo in documentation (closes issue #17395) + Reported by: pabelanger Patches: res_agi.c.patch uploaded by + pabelanger (license 224) + +2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500 + (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 + May 2010) | 2 lines Reverting patch and reopening issue #16784, + as patch breaks color display. ........ ................ + +2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 266337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 | + tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line + Only report swap on platforms which can examine those statistics + ........ + +2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | + dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines + fixes crash when creation of UDPTL fails (closes issue #17264) + Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff + uploaded by dvossel (license 671) + issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel + (license 671) Tested by: falves11 ........ + +2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com> + + * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged + revisions 266146 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010) + | 21 lines Merged revisions 266142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) + | 14 lines Use sigaction for signals which should persist past + the initial trigger, not signal. If you call signal() in a + Solaris signal handler, instead of just resetting the signal + handler, it causes the signal to refire, because the signal is + not marked as handled prior to the signal handler being called. + This effectively causes Solaris to immediately exceed the + threadstack in recursive signal handlers and crash. (closes issue + #17000) Reported by: rmcgilvr Patches: + 20100526__issue17000.diff.txt uploaded by tilghman (license 14) + Tested by: rmcgilvr ........ ................ + +2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | + dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines + fixes failed SIP Directed pickup resulting in dead channel + (closes issue #17339) Reported by: one47 Patches: + sip_magic_pickup2 uploaded by one47 (license 23) Tested by: + one47, dvossel ........ + +2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500 + (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) + | 7 lines Not finding rows in the DB does not rise to the level + of a warning. (closes issue #17062) Reported by: drookie Patches: + 20100525__issue17062.diff.txt uploaded by tilghman (license 14) + ........ ................ + + * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged + revisions 265894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 | + tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines + Construct socket name, according to the Postgres docs, and + document as such. (closes issue #17392) Reported by: dps Patches: + 20100525__issue17392.diff.txt uploaded by tilghman (license 14) + Tested by: dps ........ + +2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Recorded merge of revisions 265842 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, + 26 May 2010) | 9 lines Re-enable "always" option for videosupport + option in sip.conf. (closes issue #17016) Reported by: twilson + Patches: 17016.patch uploaded by mmichelson (license 60) Tested + by: devmod ........ + +2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | + tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines + Use configure to determine the prefixes and include directories + properly. This ensures cross-platform compatibility, even among + Linux distributions, which don't always put headers in the same + place. (closes issue #17391) Reported by: loloski ........ + +2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | + mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 + lines Properly use peer's outboundproxy for outbound REGISTERs. + The logic used in transmit_register to get the outboundproxy for + a peer was flawed since this value would be overridden shortly + afterwards when create_addr was called. In addition, this also + fixes some logic used when parsing users.conf so that the peer + name is placed in the internally-generated register string so + that an outboundproxy set in the Asterisk GUI will be used for + outbound REGISTERs. ........ + +2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com> + + * channels/chan_dahdi.c: fixes build issue with zaptel (closes + issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff + uploaded by dvossel (license 671) Tested by: aragon + +2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May + 2010) | 15 lines Merged revisions 265610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May + 2010) | 8 lines Don't mark the cdr records of unanswered queue + calls with "NOANSWER". This restores the behavior prior to + r258670. (closes issue #17334) Reported by: jvandal Patches: + queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested + by: aragon, jvandal ........ ................ + +2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com> + + * include/asterisk/options.h, main/asterisk.c, Makefile, + doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged + revisions 265320,265467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | + twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines + Add the FullyBooted AMI event It is possible to connect to the + manager interface before all Asterisk modules are loaded. To + ensure that an application does not send AMI actions that might + require a module that has not yet loaded, the application can + listen for the FullyBooted manager event. It will be sent upon + connection if all modules have been loaded, or as soon as loading + is complete. The event: Event: FullyBooted Privilege: system,all + Status: Fully Booted Review: + https://reviewboard.asterisk.org/r/639/ ........ r265467 | + twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line + Merge the rest of the FullyBooted patch ........ + +2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com> + + * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, + 24 May 2010) | 8 lines Print openh323 log to the Asterisk + console. (closes issue #17109) Reported by: under Patches: + logstream.diff uploaded by under (license 914) ........ + + * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | + mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 + lines Allow type=user SIP endpoints to be loaded properly from + realtime. (closes issue #16021) Reported by: Guggemand Patches: + realtime-type-fix.patch uploaded by Guggemand (license 897) + (altered by me slightly to avoid ref leaks) Tested by: Guggemand + ........ + +2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com> + + * main/channel.c, /: Merged revisions 265273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 | + dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines + fixes segfault when using generic plc ........ + +2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 265316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | + tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines + On systems with a LOT of RAM, a signed integer sometimes printed + negative. (closes issue #16837) Reported by: jlpedrosa Patches: + 20100504__issue16837.diff.txt uploaded by tilghman (license 14) + ........ + +2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix memory hogging behavior of app_queue. From + reviewboard: This review request is for the patch on issue 17081. + A user reported that he saw increasing numbers of allocations + stemming from app_queue.c when he would run the "queue show" CLI + command. The user reported that he was using approximately 40 + realtime queues and as he ran the CLI command more and more, the + memory usage would shoot up. As it turns out, there was a memory + leak and a separate usage of memory that, while not really a + leak, was very irresponsible. Both memory problems can be + attributed to the function init_queue(). When the "queue show" + command is run, all realtime queues have the init_queue() + function called on the in-memory queue. The idea is to place the + queue in its default state and then overwrite options specified + in the realtime backend as we read them. The first problem, the + memory leak, had to do with the fact that the string field for + the name of the first periodic announcement file was being + re-created every time init_queue was called. This patch corrects + the behavior by only calling ast_str_create if the memory has not + already been allocated. The other problem is a bit more + complicated. The majority of the strings in the call_queue + structure were changed to use the ast_string_fields API for 1.6.0 + and beyond. init_queue resets all string fields on the queue to + their default values. Then, later in the realtime queue loading + process, these string fields are set to their configured values. + For those unfamiliar with string fields, frequent resizing of a + string like this is not what the string fields API is designed + for. The result of this constant resizing is that as the queue + gets loaded, eventually space for the string runs out and so a + new memory pool, at twice the size of the previously allocated + one, is created for the string fields. The reporter of issue + 17081 wrote a script that ran the "queue show" CLI command 2100 + times. By the end, each of his 40 queues was taking about a + megabyte of memory apiece just for their string fields. My fix + for this problem is to revert the call_queue structure from using + string fields. In my patch here, I have moved the queue back to + using fixed-sized buffers. I ran the script provided by the + reporter of 17081 and determined that I no longer saw the + steadily-increasing memory usage that I had seen before applying + the patch. (closes issue #17081) Reported by: wliegel Patches: + 17081v2.patch uploaded by mmichelson (license 60) Tested by: + wliegel, mmichelson Review: + https://reviewboard.asterisk.org/r/651/ + + * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions + 265090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May + 2010) | 15 lines Merged revisions 265089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May + 2010) | 8 lines Don't hang up on a queue caller if the file we + attempt to play does not exist. This also fixes a documentation + mistake in file.h that made my original attempt to correct this + problem not work correctly. (closes issue #17061) Reported by: + RoadKill ........ ................ + + * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | + mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 + lines Be sure to set the sin_family on the proxy when allocating. + (closes issue #17157) Reported by: stuarth ........ + + * /, include/asterisk/channel.h: Merged revisions 265000 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 + (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, + 21 May 2010) | 3 lines Fix grammatical error in comment. ........ + ................ + + * main/channel.c, main/autoservice.c, /, + include/asterisk/channel.h: Merged revisions 264997 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 + (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May + 2010) | 32 lines Allow ast_safe_sleep to defer specific frames + until after the sleep has concluded. From reviewboard Background: + A Digium customer discovered a somewhat odd bug. The setup is + that parties A and B are bridged, and party A places party B on + hold. While party B is listening to hold music, he mashes a bunch + of DTMF. Party A takes party B off hold while this is happening, + but party B continues to hear hold music. I could reproduce this + about 1 in 5 times. The issue: When DTMF features are enabled and + a user presses keys, the channel that the DTMF is streamed to is + placed in an ast_safe_sleep for 100 ms, the duration of the + emulated tone. If an AST_CONTROL_UNHOLD frame is read from the + channel during the sleep, the frame is dropped. Thus the unhold + indication is never made to the channel that was originally + placed on hold. The fix: Originally, I discussed with Kevin + possible ways of fixing the specific problem reported. However, + we determined that the same type of problem could happen in other + situations where ast_safe_sleep() is used. Using autoservice as a + model, I modified ast_safe_sleep_conditional() to defer specific + frame types so they can be re-queued once the sleep has finished. + I made a common function for determining if a frame should be + deferred so that there are not two identical switch blocks to + maintain. Review: https://reviewboard.asterisk.org/r/674/ + ........ ................ + +2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com> + + * /, main/callerid.c: Merged revisions 264828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010) + | 13 lines Merged revisions 264820 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) + | 6 lines ast_callerid_parse() had a path that left name + uninitialized. Several callers of ast_callerid_parse() do not + initialize the name parameter before calling thus there is the + potential to use an uninitialized pointer. ........ + ................ + +2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 264779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 | + tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines + Let ExtensionState resolve dynamic hints. (closes issue #16623) + Reported by: tilghman Patches: 20100116__issue16623.diff.txt + uploaded by tilghman (license 14) Tested by: lmadsen ........ + + * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 | + tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines + Error message fix. (closes issue #17356) Reported by: kenner + Patches: app_stack.c.diff uploaded by kenner (license 1040) + ........ + +2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/_private.h, include/asterisk/options.h, + main/asterisk.c, main/loader.c, main/channel.c, /, + channels/chan_sip.c: Merged revisions 264452 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | + mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 + lines Fix transcode_via_sln option with SIP calls and improve PLC + usage. From reviewboard: The problem here is a bit complex, so + try to bear with me... It was noticed by a Digium customer that + generic PLC (as configured in codecs.conf) did not appear to + actually be having any sort of benefit when packet loss was + introduced on an RTP stream. I reproduced this issue myself by + streaming a file across an RTP stream and dropping approx. 5% of + the RTP packets. I saw no real difference between when PLC was + enabled or disabled when using wireshark to analyze the RTP + streams. After analyzing what was going on, it became clear that + one of the problems faced was that when running my tests, the + translation paths were being set up in such a way that PLC could + not possibly work as expected. To illustrate, if packets are lost + on channel A's read stream, then we expect that PLC will be + applied to channel B's write stream. The problem is that generic + PLC can only be done when there is a translation path that moves + from some codec to SLINEAR. When I would run my tests, I found + that every single time, read and write translation paths would be + set up on channel A instead of channel B. There appeared to be no + real way to predict which channel the translation paths would be + set up on. This is where Kevin swooped in to let me know about + the transcode_via_sln option in asterisk.conf. It is supposed to + work by placing a read translation path on both channels from the + channel's rawreadformat to SLINEAR. It also will place a write + translation path on both channels from SLINEAR to the channel's + rawwriteformat. Using this option allows one to predictably set + up translation paths on all channels. There are two problems with + this, though. First and foremost, the transcode_via_sln option + did not appear to be working properly when I was placing a SIP + call between two endpoints which did not share any common + formats. Second, even if this option were to work, for PLC to be + applied, there had to be a write translation path that would go + from some format to SLINEAR. It would not work properly if the + starting format of translation was SLINEAR. The one-line change + presented in this review request in chan_sip.c fixed the first + issue for me. The problem was that in sip_request_call, the + jointcapability of the outbound channel was being set to the + format passed to sip_request_call. This is nativeformats of the + inbound channel. Because of this, when + ast_channel_make_compatible was called by app_dial, both channels + already had compatibly read and write formats. Thus, no + translation path was set up at the time. My change is to set the + jointcapability of the sip_pvt created during sip_request_call to + the intersection of the inbound channel's nativeformats and the + configured peer capability that we determined during the earlier + call to create_addr. Doing this got the translation paths set up + as expected when using transcode_via_sln. The changes presented + in channel.c fixed the second issue for me. First and foremost, + when Asterisk is started, we'll read codecs.conf to see the value + of the genericplc option. If this option is set, and ast_write is + called for a frame with no data, then we will attempt to fill in + the missing samples for the frame. The implementation uses a + channel datastore for maintaining the PLC state and for creating + a buffer to store PLC samples in. Even when we receive a frame + with data, we'll call plc_rx so that the PLC state will have + knowledge of the previous voice frame, which it can use as a + basis for when it comes time to actually do a PLC fill-in. So, + reviewers, now I ask for your help. First off, there's the one + line change in chan_sip that I have put in. Is it right? By my + logic it seems correct, but I'm sure someone can tell me why it + is not going to work. This is probably the change I'm least + concerned about, though. What concerns me much more is the set of + changes in channel.c. First off, am I even doing it right? When I + run tests, I can clearly see that when PLC is activated, I see a + significant increase in RTP traffic where I would expect it to + be. However, in my humble opinion, the audio sounds kind of + crappy whenever the PLC fill-in is done. It sounds worse to me + than when no PLC is used at all. I need someone to review the + logic I have used to be sure that I'm not misusing anything. As + far as I can see my pointer arithmetic is correct, and my use of + AST_FRIENDLY_OFFSET should be correct as well, but I'm sure + someone can point out somewhere where I've done something + incorrectly. As I was writing this review request up, I decided + to give the code a test run under valgrind, and I find that for + some reason, calls to plc_rx are causing some invalid reads. + Apparently I'm reading past the end of a buffer somehow. I'll + have to dig around a bit to see why that is the case. If it's + obvious to someone reviewing, speak up! Finally, I have one other + proposal that is not reflected in my code review. Since without + transcode_via_sln set, one cannot predict or control where a + translation path will be up, it seems to me that the current + practice of using PLC only when transcoding to SLINEAR is not + useful. I recommend that once it has been determined that the + method used in this code review is correct and works as expected, + then the code in translate.c that invokes PLC should be removed. + Review: https://reviewboard.asterisk.org/r/622/ ........ + +2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com> + + * main/udptl.c, /: Merged revisions 264400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 | + dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines + fixes infinite loop during udptl.c's decode_open_type When + decode_length returns the length there is a check to see if that + length is negative, if so the decode loop breaks as this means + the limit has been reached. The problem here is that length is an + unsigned int, so length can never be negative. This resulted in + an infinite loop. (issue #17352) ........ + +2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com> + + * main/udptl.c, /: Merged revisions 264379 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 | + mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4 + lines Cast an unsigned int to a signed int when comparing it with + 0. (AST-377) ........ + + * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500 + (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May + 2010) | 5 lines Set quieted flag when receiving a dtmf tone + during playback in speechbackground. (closes issue #16966) + Reported by: asackheim ........ ................ + +2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | + dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines + fixes crash in check_rtp_timeout During deadlock avoidance the + sip dialog pvt is locked and unlocked. When this occurs we have + no guarantee the pvt's owner is still valid. We were trying to + access the pvt's owner after this without checking to see if it + still existed first. (closes issue #17271) Reported by: under + Patches: check_rtp_timeout.diff uploaded by under (license 914) + Tested by: dvossel ........ + +2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/options.h, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 264249 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) + | 24 lines Merged revisions 264248 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) + | 17 lines Internal timing is now on by default, if you're using + DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is + that this version ensures that a timer is always available, + whereas in previous versions, it was possible for DAHDI to be + loaded, but have no drivers to actually generate timing. If + internal_timing was turned on in this circumstance, a complete + lack of audio would result. This is the reason why + internal_timing was not on by default. However, now that DAHDI + ensures the availability of a timer, there is no reason for this + setting to be off (and in fact, it solves a great many initial + user problems). (closes issue #15932) Reported by: dimas Patches: + 20100519__issue15932.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + + * main/dsp.c, /: Merged revisions 264204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 | + tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines + Keep track of digit duration, when we're decoding inband to pass + DTMF frames. (closes issue #17235) Reported by: frawd Patches: + new_dtmf_dsp_len.patch uploaded by frawd (license 610) + 20100518__issue17235.diff.txt uploaded by tilghman (license 14) + Tested by: frawd ........ + +2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com> + + * main/rtp.c, /: Merged revisions 264114 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 | + dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines + fixes crash during dtmf During the processing of Cisco dtmf the + dtmf samples were not being calculated correctly. In an attempt + to determine what sample rate was being used, a NULL frame was + processed which caused a crash. This patch resolves this. (closes + issue #17248) Reported by: falves11 Patches: issue_17248.diff + uploaded by dvossel (license 671) ........ + +2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz> + + * /, configs/indications.conf.sample: Merged revisions 264031 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19 + May 2010) | 8 lines fix incorrectly typed indications for [nz] + stutter and dialrecall (closes issue #17359) Reported by: + alecdavis Patches: bug17359.diff.txt uploaded by alecdavis + (license 585) ........ + +2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com> + + * main/dsp.c, /: Merged revisions 263950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010) + | 15 lines Merged revisions 263949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) + | 8 lines Because progress is called multiple times, across + several frames, we must persist states when detecting multitone + sequences. (closes issue #16749) Reported by: dant Patches: + dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: + dant ........ ................ + +2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com> + + * main/strings.c, /: Merged revisions 263904 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 | + dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines + fixes segfault on logging (closes issue #17331) Reported by: + under Patches: utils.diff uploaded by under (license 914) + segfault_on_logging.diff uploaded by dvossel (license 671) Tested + by: under, dvossel ........ + +2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com> + + * apps/app_directory.c, /: Merged revisions 263807 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500 + (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) + | 10 lines Modify directory name reading to be interrupted with + operator or pound escape. In the case of accidentally entering + the wrong first three letters for the reading, users could be + very frustrated if the name listing is very long. This allows + interrupting the reading by pressing 0 or #. 0 will attempt to + execute a configured operator (o) extension and # will exit and + proceed in the dialplan. ABE-2200 ........ ................ + +2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com> + + * /, main/devicestate.c: Merged revisions 263640 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May + 2010) | 16 lines Merged revisions 263639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May + 2010) | 10 lines Fix logic error when checking for a devstate + provider. When using strsep, if one of the list of specified + separators is not found, it is the first parameter to strsep + which is now NULL, not the pointer returned by strsep. This issue + isn't especially severe in that the worst it is likely to do is + waste some cycles when a device with no '/' and no ':' is passed + to ast_device_state. ........ ................ + +2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010) + | 9 lines With IMAP backend, messages in INBOX were counted twice + for MWI. (closes issue #17135) Reported by: edhorton Patches: + 20100513__issue17135.diff.txt uploaded by tilghman (license 14) + 17135_2.diff uploaded by ebroad (license 878) Tested by: + edhorton, ebroad ........ + + * main/app.c: Don't close 'n', just close 'above_n'. (closes issue + #17345) Reported by: wdoekes + +2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com> + + * main/manager.c, /: Merged revisions 263457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010) + | 19 lines Recorded merge of revisions 263456 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) + | 11 lines Manager cookies are not compatible with RFC2109. The + Version field in the cookies we're setting contain quotes around + the version number which is not compatible with RFC2109 and + breaks some implementations. (closes issue #17231) Reported by: + ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by + ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by + ecarruda (license 559) Tested by: ecarruda, russell ........ + ................ + + * sounds/Makefile, /: Merged revisions 263375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010) + | 16 lines Merged revisions 263374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) + | 8 lines Update link to new version of core sounds. The latest + version of the core sounds files 1.4.19 now includes the missing + queue-minute sound file which is called by app_queue but which + has been missing. (closes issue #17123) Reported by: n8ideas + ........ ................ + +2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com> + + * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer + policy dialstring option + +2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com> + + * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 | + tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines + Make the Makefile logic more explicit and move the Snow Leopard + logic down to where it's not executed on non-Darwin systems. + (closes issue #17028) Reported by: pabelanger Patches: + issue17028_20100315.patch uploaded by seanbright (license 71) + 20100315__issue17028.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman, pabelanger ........ + +2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010) + | 1 line Fix inverted logic in cli command: ss7 set debug on/off + ........ + +2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com> + + * channels/chan_console.c, /: Merged revisions 262897 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010) + | 4 lines Fix an off by one error that causes a crash. Thanks to + Raymond Burke for pointing it out. ........ + +2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com> + + * main/loader.c, main/cli.c, /: Merged revisions 262800 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed, + 12 May 2010) | 8 lines Notify CLI when modules is loaded / + unloaded (closes issue #17308) Reported by: pabelanger Patches: + cli.modules.patch uploaded by pabelanger (license 224) Tested by: + pabelanger, russell ........ + +2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com> + + * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 | + lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines + Revert previous WARNING message removal. Marquis42 suggested a + better method of doing what I wanted because I ended up removing + the WARNING message for all instances when really I just wanted + to remove it for the 'return' keyword, not everything. (issue + #17145) ........ + + * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 | + lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines + Remove unnecessary WARNING message in ael/pval.c (closes issue + #17145) Reported by: okrief ........ + +2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com> + + * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010) + | 17 lines Merged revisions 262662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) + | 11 lines fixes app_meetme dsp error We attempted to detect + silence after translating a frame from signed linear. This caused + a flooding of errors. To resolve this the code to detect silence + was moved before the translation. (closes issue #17133) Reported + by: jsdyer ........ ................ + +2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 | + tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines + Ensure the arguments are initialized. Also miscellaneous CG + cleanup. (closes issue #16576) Reported by: uxbod Patches: + 20100505__issue16576.diff.txt uploaded by tilghman (license 14) + Tested by: uxbod ........ + + * /, include/asterisk/causes.h: Merged revisions 262513 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 + May 2010) | 7 lines Move cause 200 to cause 26, as specified in + Q.850. Also cleanup the formatting and add a few more that seem + like good candidates. (closes issue #16157) Reported by: wimpy + ........ + +2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com> + + * /, res/Makefile: Merged revisions 262422 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) | + 18 lines Merged revisions 262421 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | + 11 lines Use a less silly method for modifying a flex-generated + file. The sed syntax that was used wasn't actually valid, causing + some versions to choke. This is the method that is used in 1.6.x+ + for similar changes. (closes issue #16696) Reported by: bklang + Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested + by: qwell ........ ................ + +2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com> + + * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 | + pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8 + lines Improve logging by displaying line number (closes issue + #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded + by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger + ........ + + * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 | + pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 + lines Improve logging information for misconfigured contexts + (closes issue #17238) Reported by: pprindeville Patches: + chan_sip-bug17238.patch uploaded by pprindeville (license 347) + Tested by: pprindeville ........ + +2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500 + (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 + May 2010) | 2 lines Fix issue #17302 a slightly different way + (mad props to Qwell) ........ ................ + +2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com> + + * /, apps/app_directed_pickup.c: Merged revisions 262240 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10 + May 2010) | 9 lines fixes PickupChan application (closes issue + #16863) Reported by: schern Patches: app_directed_pickup.c.patch + uploaded by schern (license 995) for_trunk.diff uploaded by + cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, + rickead2000, dvossel ........ + + * channels/chan_console.c, /: Merged revisions 262236 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) + | 11 lines fixes crash in chan_console There is a race condition + between console_hangup() and start_stream(). It is possible for + console_hangup() to be called and then the stream thread to begin + after the hangup. To avoid this a check in start_stream() to make + sure the pvt-owner still exists while the pvt lock is held is + made. If the owner is gone that means the channel hung up and + start_stream should be aborted. ........ + +2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com> + + * /, Makefile.rules: Merged revisions 262152 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010) + | 17 lines Merged revisions 262151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) + | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes + issue #17297) Reported by: jcovert Patches: + 20100506__issue17297.diff.txt uploaded by tilghman (license 14) + (closes issue #17302) Reported by: jcovert ........ + ................ + +2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com> + + * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4, + autoconf/ast_c_define_check.m4, /, configure, + include/asterisk/autoconfig.h.in: Merged revisions 262102 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 + May 2010) | 5 lines Cleanup a bit more by getting rid of useless + version defines. Also make library detection use passed CFLAGS. + (closes issue #17309) Reported by: stuarth ........ + + * /, configure, configure.ac: Merged revisions 262048 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010) + | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only + ........ + + * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 | + tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines + Double free crash (closes issue #17245) Reported by: + thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by + tilghman (license 14) Tested by: murraytm ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 261913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 | + tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 + lines Use the detected pthread building flags in every place, + instead of hardcoding -lpthread. We nicely detect the right flags + on each system for building Asterisk with pthreads, then ignore + it for every other build option that requires us to build with + pthreads. This caused some items to return a false negative. Also + cleanup some minor naming issues that caused "library library" + redundancy in the output. (closes issue #17303) Reported by: + stuarth Patches: 20100507__issue17303.diff.txt uploaded by + tilghman (license 14) Tested by: stuarth ........ + +2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com> + + * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 | + lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines + Update UPGRADE-1.6.txt stating insecure=very has been removed. + (closes issue #17282) Reported by: stuarth Tested by: stuarth + ........ + +2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500 + (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) + | 8 lines Only allow the operator key to be accepted after + leaving a voicemail. Or rather disallow the operator key from + being accepted when not offered, such as after finishing a + recording from within the mailbox options menu. ABE-2121 SWP-1267 + ........ ................ + +2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com> + + * sounds/Makefile, /: Merged revisions 261609 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) | + 11 lines Merged revisions 261608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | + 4 lines Use the versioned MOH tarballs, now that we have them. + This makes for more reproducibility. Prompted by a discussion in + #asterisk-dev ........ ................ + +2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 | + tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines + Permit more lines within a SIP body to be parsed. The example + given within the related issue showed 120 lines, which was mostly + a result of the body being XML. (closes issue #17179) Reported + by: khw ........ + +2010-06-01 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.8 Released. + +2010-05-26 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.8-rc2 Released. + +2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com> + + * Merged r265610 from 1.4: + + Don't mark the cdr records of unanswered queue calls with "NOANSWER". + This restores the behavior prior to r258670. + + (closes issue #17334) + Reported by: jvandal + Patches: + queue-cdr-fixes1.diff uploaded by mnicholson (license 96) + Tested by: aragon, jvandal + +2010-05-06 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.8-rc1 Released + +2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com> + + * tests/test_heap.c: Add test case that ensures the heap handles + arbitrary removals properly. (issue #17277) Reported by: + cappucinoking Patches: test_heap.diff uploaded by cappucinoking + (license 1036) Tested by: cappucinoking, russell + + * /, main/heap.c: Merged revisions 261496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 | + russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines + Fix handling of removing nodes from the middle of a heap. This + bug surfaced in 1.6.2 and does not affect code in any other + released version of Asterisk. It manifested itself as SIP qualify + not happening when it should, causing peers to go unreachable. + This was debugged down to scheduler entries sometimes not getting + executed when they were supposed to, which was in turn caused by + an error in the heap code. The problem only sometimes occurs, and + it is due to the logic for removing an entry in the heap from an + arbitrary location (not just popping off the top). The scheduler + performs this operation frequently when entries are removed + before they run (when ast_sched_del() is used). In a normal pop + off of the top of the heap, a node is taken off the bottom, + placed at the top, and then bubbled down until the max heap + property is restored (see max_heapify()). This same logic was + used for removing an arbitrary node from the middle of the heap. + Unfortunately, that logic is full of fail. This patch fixes that + by fully restoring the max heap property when a node is thrown + into the middle of the heap. Instead of just pushing it down as + appropriate, it first pushes it up as high as it will go, and + _then_ pushes it down. Lastly, fix a minor problem in + ast_heap_verify(), which is only used for debugging. If a parent + and child node have the same value, that is not an error. The + only error is if a parent's value is less than its children. A + huge thanks goes out to cappucinoking for debugging this down to + the scheduler, and then producing an ast_heap test case that + demonstrated the breakage. That made it very easy for me to focus + on the heap logic and produce a fix. Open source projects are + awesome. (closes issue #16936) Reported by: ib2 Tested by: + cappucinoking, crjw (closes issue #17277) Reported by: + cappucinoking Patches: heap-fix.rev2.diff uploaded by russell + (license 2) Tested by: cappucinoking, russell ........ + +2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | + 4 lines When failing to configure, don't destroy 'cfg' twice + Fixes a crash when some config section had an incorrect channel + config. ........ + +2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com> + + * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May + 2010) | 19 lines Merged revisions 261274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May + 2010) | 12 lines Registration fix for SIP realtime. Make sure + realtime fields are not empty. (closes issue #17266) Reported by: + Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick + Lewis (license 657) Tested by: Nick_Lewis, sberney Review: + https://reviewboard.asterisk.org/r/643/ ........ ................ + + * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 | + pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10 + lines 'queue reset stats' erroneously clears wrapuptime + configuration. Resets each member's lastcall to 0 now. (closes + issue #17262, #16519) Reported by: rain Patches: + wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested + by: rain ........ + +2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 261095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) + | 18 lines Merged revisions 261093-261094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) + | 7 lines Protect against overflow, when calculating how long to + wait for a frame. (closes issue #17128) Reported by: under + Patches: d.diff uploaded by under (license 914) ........ r261094 + | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 + lines Add a tiny corner case to the previous commit ........ + ................ + +2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500 + (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) + | 12 lines Voicemail transfer to operator should occur + immediately, not after main menu. There were two scenarios in the + advanced options that while using the operator=yes and review=yes + options, the transfer occurred only after exiting the main menu + (after sending a reply or leaving a message for an extension). + Now after the audio is processed for the reply or message the + transfer occurs immediately as expected. ABE-2107 ABE-2108 + ........ ................ + +2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com> + + * configs/sip.conf.sample, include/asterisk/frame.h, + main/channel.c, /, channels/chan_sip.c: Merged revisions 254450 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 + Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that + arrive before a T.38-capable application is executing on a + channel. This patch addresses an issue found during working with + end-users using res_fax. If an incoming call is answered in the + dialplan, or jumps to the 'fax' extension due to reception of a + CNG tone (with faxdetect enabled), and then the remote endpoint + sends a T.38 re-INVITE, it is possible for the channel's T.38 + state to be 'T38_STATE_NEGOTIATING' when the application starts + up. Unfortunately, even if the application wants to use T.38, it + can't respond to the peer's negotiation request, because the + AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent + originally has been lost, and the application needs the content + of that frame to be able to formulate a reply. This patch adds a + new 'request' type to AST_CONTROL_T38_PARAMETERS, + AST_T38_REQUEST_PARMS. If the application sends this request, + chan_sip will re-send the original control frame (with + AST_T38_REQUEST_NEGOTIATE as the request type), and the + application can respond as normal. If this occurs within the five + second timeout in chan_sip, the automatic cancellation of the + peer reinvite will be stopped, and the application will 'own' the + negotiation process from that point onwards. This also improves + the code path in chan_sip to allow sip_indicate(), when called + for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero + response, which should have been in place before since the + control frame *can* fail to be processed properly. It also + modifies ast_indicate() to return whatever result the channel + driver returned for this control frame, rather than converting + all non-zero results into '-1'. Finally, the new request type + intentionally returns a positive value, so that an application + that sends AST_T38_REQUEST_PARMS can know for certain whether the + channel driver accepted it and will be replying with a control + frame of its own, or whether it was ignored (if the + sip_indicate()/ast_indicate() path had properly supported failure + responses before, this would not be necessary). This patch also + modifies res_fax to take advantage of the new request. In + addition, this patch makes sip_t38_abort() actually lock the + private structure before doing its work... bad programmer, no + donut. This patch also enhances chan_sip's 'faxdetect' support to + allow triggering on T.38 re-INVITEs received as well as CNG tone + detection. Review: https://reviewboard.asterisk.org/r/556/ + ........ + +2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com> + + * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260802 | qwell | 2010-05-04 10:49:57 -0500 + (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May + 2010) | 1 line Fix fallout from removing from configure script. + Pointed out by philipp64 on #asterisk-dev ........ + ................ + + * /: Fix merge props + +2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com> + + * Makefile, /: Merged revisions 260661-260662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May + 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend + libdir when executing mkpkgconfig allowing non-root installs to + work. (closes issue #17268) Reported by: pabelanger Patches: + issue17268.patch uploaded by pabelanger (license 224) Tested by: + pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 + -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ + part. Thanks Qwell. ........ + +2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com> + + * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500 + (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 + May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. + ........ ................ + +2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 + (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) + | 11 lines Ensure channel state is not incorrectly set in the + case of a very early answer. The needringing bit was being read + in dahdi_read after answering thereby setting the state to + ringing from up. This clears needringing upon answering so that + is no longer possible. (closes issue #17067) Reported by: tzafrir + Patches: needringing.diff uploaded by tzafrir (license 46) + ........ ................ + +2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500 + (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr + 2010) | 18 lines Fix potential crash from race condition due to + accessing channel data without the channel locked. In + res_musiconhold.c, there are several places where a channel's + stream's existence is checked prior to calling ast_closestream on + it. The issue here is that in several cases, the channel was not + locked while checking the stream. The result was that if two + threads checked the state of the channel's stream at + approximately the same time, then there could be a situation + where both threads attempt to call ast_closestream on the + channel's stream. The result here is that the refcount for the + stream would go below 0, resulting in a crash. I have added + proper channel locking to res_musiconhold.c to ensure that we do + not try to check chan->stream without the channel locked. A + Digium customer has been using this patch for several weeks and + has not had any crashes since applying the patch. ABE-2147 + ........ ................ + +2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: Merged revisions 260292 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 | + tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13 + lines Don't allow file descriptors to go above 64k, when we're + closing them in a fork(2). This saves time, when, even though the + system allows the process limit to be that high, the practical + limit is much lower. (closes issue #17223) Reported by: + dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by + tilghman (license 14) Tested by: dbackeberg ........ + + * configs/extensions.conf.sample, /: Merged revisions 260280 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30 + Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan + context. (closes issue #17263) Reported by: pprindeville Patches: + freenum-dialplan.patch#3 uploaded by pprindeville (license 347) + ........ + +2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 + (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) + | 26 lines DTMF CallerID detection problems. The code handling + DTMF CallerID drops digits on long CallerID numbers and may + timeout waiting for the first ring with shorter numbers. The DTMF + emulation mode was not turned off when processing DTMF CallerID. + When the emulation code gets behind in processing the DTMF digits + it can skip a digit. For shorter numbers, the timeout may have + been too short. I increased it from 2 seconds to 4 seconds. Four + seconds is a typical time between rings for many countries. + (closes issue #16460) Reported by: sum Patches: issue16460.patch + uploaded by rmudgett (license 664) issue16460_v1.6.2.patch + uploaded by rmudgett (license 664) Tested by: sum, rmudgett + Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA + AST-334 JIRA SWP-901 ........ ................ + +2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 260148 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 + Apr 2010) | 2 lines Pattern match fail. ........ + +2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com> + + * main/audiohook.c, /, include/asterisk/audiohook.h: Merged + revisions 260050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) + | 21 lines Merged revisions 260049 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) + | 14 lines Fixes crash in audiohook_write_list The middle_frame + in the audiohook_write_list function was being freed if a + audiohook manipulator returned a failure. This is incorrect + logic. This patch resolves this and adds detailed descriptions of + how this function should work and why manipulator failures must + be ignored. (closes issue #17052) Reported by: dvossel Tested by: + dvossel (closes issue #16196) Reported by: atis Review: + https://reviewboard.asterisk.org/r/623/ ........ ................ + +2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | + mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 + lines Don't override peer context with domain context. (closes + issue #17040) Reported by: pprindeville Patches: + asterisk-1.6-bugid17040.patch uploaded by pprindeville (license + 347) Tested by: pprindeville Review: + https://reviewboard.asterisk.org/r/565/ ........ + +2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com> + + * main/channel.c, channels/chan_local.c, /: Merged revisions 259870 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 + (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) + | 33 lines resolves deadlocks in chan_local Issue_1. In the + local_hangup() 3 locks must be held at the same time... pvt, + pvt->chan, and pvt->owner. Proper deadlock avoidance is done when + the channel to hangup is the outbound chan_local channel, but + when it is not the outbound channel we have an issue... We + attempt to do deadlock avoidance only on the tech pvt, when both + the tech pvt and the pvt->owner are locked coming into that loop. + By never giving up the pvt->owner channel deadlock avoidance is + not entirely possible. This patch resolves that by doing deadlock + avoidance on both the pvt->owner and the pvt when trying to get + the pvt->chan lock. Issue_2. ast_prod() is used in + ast_activate_generator() to queue a frame on the channel and make + the channel's read function get called. This function is used in + ast_activate_generator() while the channel is locked, which + mean's the channel will have a lock both from the generator code + and the frame_queue code by the time it gets to chan_local.c's + local_queue_frame code... local_queue_frame contains some of the + same crazy deadlock avoidance that local_hangup requires, and + this recursive lock prevents that deadlock avoidance from + happening correctly. This patch removes ast_prod() from the + channel lock so only one lock is held during the + local_queue_frame function. (closes issue #17185) Reported by: + schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel + (license 671) issue_17185_v2.diff uploaded by dvossel (license + 671) Tested by: schmoozecom, GameGamer43 Review: + https://reviewboard.asterisk.org/r/631/ ........ ................ + +2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com> + + * config.guess: Merged revisions 259853 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) + | 14 lines Merged revisions 259852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) + | 6 lines Update config.guess. Updating config.guess because + after installing Ubuntu Server 9.10 and running all the update + scripts, running ./configure would not continue because it was + unable to determine what kind of system I had. After updating + config.guess things started working again. ........ + ................ + +2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac: Merged revisions 259848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259848 | qwell | 2010-04-28 15:32:14 -0500 + (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr + 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so + systems without install can use install-sh from our source dir. + ........ ................ + + * makeopts.in, /: Merged revisions 259837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | + 9 lines Merged revisions 259833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | + 1 line Missed this when removing $ID ........ ................ + + * Makefile, /, configure, configure.ac: Merged revisions 259760 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259760 | qwell | 2010-04-28 14:19:54 -0500 + (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | + 7 lines Remove usage of `id` since it isn't useful and was + causing breakge. Solaris `id` doesn't support the -u argument. + Instead of figuring out how to fix this to work on Solaris, I + decided to check why it was necessary and where else it was used. + It was only used in one place, and it hasn't been needed for a + very long time (I question whether it was ever needed). ........ + ................ + +2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500 + (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) + | 4 lines Do not play goodbye prompt after timeout of message + review. ABE-2124 ........ ................ + +2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 + (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) + | 11 lines DAHDI "WARNING" message is confusing and vague + "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed + failed: Success" Changed the warning to "Failed to decode + CallerID on channel 'name'". The message before it is likely more + specific about why the CallerID decode failed. SWP-501 AST-283 + ........ ................ + +2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com> + + * sounds/Makefile: Merged revisions 259527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) + | 23 lines Merged revisions 259526 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) + | 15 lines Update sounds files. * Add additional sounds prompts + for say_enumeration * Update the English conference sounds + prompts so they are better quality and all sound more consistent + * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files + to include all present sound files Both core (en, fr, es) and + extra (en, fr) sounds files have been updated. (closes issue + #16200) Reported by: murf (closes issue #17137) Reported by: + lmadsen ........ ................ + +2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com> + + * main/editline/configure.in, /, main/editline/configure, + main/editline/Makefile.in: Merged revisions 259439 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | + 5 lines Add gar to the check for AR for those silly OSes + (Solaris) that don't have ar. autoconf2.13 couldn't handle + AC_PROG_GREP, so I removed it. This is fine, since we don't need + to use anything that the configure script doesn't. ........ + + * /: Unblock revision 259439. + + * /, configure, configure.ac: Merged revisions 259353 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259353 | qwell | 2010-04-27 14:31:55 -0500 + (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | + 5 lines Support the silly OSes that don't have ar and strip. + Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't + specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to + AC_CHECK_TOOLS. ........ ................ + +2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 259307 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) + | 21 lines Merged revisions 259270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) + | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue + #7321 implements a new chan_dahdi configuration option. However, + a change mentioned in the issue was never implemented. This is + the change that will allow the feature to work. I added a note to + chan_dahdi.conf.sample about the feature. (closes issue #17143) + Reported by: djensen99 Patches: diff.txt uploaded by djensen99 + (license NA) (One line change) Tested by: djensen99 ........ + ................ + +2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 259105 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr + 2010) | 9 lines Merged revisions 259104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr + 2010) | 3 lines Let compilation succeed warning-free when + DONT_OPTIMIZE is turned off. ........ ................ + + * main/channel.c, /: Merged revisions 259023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr + 2010) | 19 lines Merged revisions 259018 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr + 2010) | 13 lines Prevent Newchannel manager events for dummy + channels. No Newchannel manager event will be fired for channels + that are allocated to not match a registered technology type. + Thus bogus channels allocated solely for variable substitution or + CDR operations do not result in a Newchannel event. (closes issue + #16957) Reported by: atis Review: + https://reviewboard.asterisk.org/r/601 ........ ................ + +2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com> + + * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 | + lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines + Small error in the T.140 RTP port verbose log. (closes issue + #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff + uploaded by frawd (license 610) Tested by: russell ........ + +2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com> + + * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) + | 13 lines Merged revisions 258775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) + | 6 lines When StopMonitor is called, ensure that it will not be + restarted by a channel event. (closes issue #16590) Reported by: + kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm + (license 888) ........ ................ + +2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com> + + * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions + 258671,258675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr + 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May + 2009) | 8 lines Set the proper disposition on originated calls. + (closes issue #14167) Reported by: jpt Patches: + call-file-missing-cdr2.diff uploaded by mnicholson (license 96) + Tested by: dlotina, rmartinez, mnicholson ........ r258670 | + mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 + lines Fix broken CDR behavior. This change allows a CDR record + previously marked with disposition ANSWERED to be set as BUSY or + NO ANSWER. Additionally this change partially reverts r235635 and + does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated + from ast_call(). To preserve proper CDR behavior, the + AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in + ast_bridge_call(). (closes issue #16797) Reported by: + VarnishedOtter Tested by: mnicholson ........ (closes issue + #16222) Reported by: telles Tested by: mnicholson + ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500 + (Thu, 22 Apr 2010) | 2 lines Fix previous commit. + ................ + +2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com> + + * /, main/event.c: Merged revisions 258632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only + merge the bug fixes, not the unit test. ........ r258632 | + russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines + Add ast_event subscription unit test and fix some ast_event API + bugs. This patch introduces another test in test_event.c that + exercises most of the subscription related ast_event API calls. I + made some minor additions to the existing event allocation test + to increase API coverage by the test code. Finally, I made a list + in a comment of API calls not yet touched by the test module as a + to-do list for future test development. During the development of + this test code, I discovered a number of bugs in the event API. + 1) subscriptions to AST_EVENT_ALL were not handled appropriately + in a couple of different places. The API allows a subscription to + all event types, but with IE parameters, just as if it was a + subscription to a specific event type. However, the parameters + were being ignored. This affected ast_event_check_subscriber() + and event distribution to subscribers. 2) Some of the logic in + ast_event_check_subscriber() for checking subscriptions against + query parameters was wrong. Review: + https://reviewboard.asterisk.org/r/617/ ........ + + * /, doc/tex/channelvariables.tex: Merged revisions 258515 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22 + Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........ + +2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500 + (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) + | 8 lines Fix looping forever when no input received in certain + voicemail menu scenarios. Specifically, prompting for an + extension (when leaving or forwarding a message) or when + prompting for a digit (when saving a message or changing + folders). ABE-2122 SWP-1268 ........ ................ + +2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com> + + * doc/tex/asterisk.tex: Remove missed line in previous merge. + (issue #17220) + + * configure: Forgot to merge the updated configure script. (issue + #17220) + + * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in, + doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile, + configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex, + build_tools/prep_tarball: Merged revisions 258351 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010) + | 20 lines Add ability to generate ASCII documentation from the + TeX files. These changes add the ability to run 'make + asterisk.txt' just like the existing 'make asterisk.pdf' commands + to generate a text document from the TeX files we have in the + doc/tex/ directory. I've also updated a few of the .tex files + because they weren't properly escaping certain characters so they + would show up as Unicode characters (like [U+021C]). Made changes + to the configure scripts so it would detect the catdvi program + which is required to convert the .dvi file generated by latex. + I've also added a few lines to the build_tools/prep_tarball + script so that the text documentation gets generated and added to + future tarballs of Asterisk releases. (closes issue #17220) + Reported by: lmadsen Patches: asterisk.txt.patch uploaded by + lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger + (license 224) Tested by: lmadsen, pabelanger ........ + +2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | + dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines + fixes issue with double "sip:" in header field This is a clear + mistake in logic. Future discussions about how to avoid having to + handle uri's like this should take place in the future, but this + fix needs to go in for now. (closes issue #15847) Reported by: + ebroad Patches: doublesip.patch uploaded by ebroad (license 878) + ........ + +2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com> + + * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20 + Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on + asterisk-users. ........ + + * configs/extensions.conf.sample, /: Merged revisions 258147 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20 + Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers + (http://www.freenum.org). Minor tweaks and documentation added by + me. (closes issue #17058) Reported by: pprindeville Patches: + freenum.patch#5 uploaded by pprindeville (license 347) Tested by: + lmadsen ........ + +2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500 + (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) + | 11 lines Play correct prompt when voicemail store failure + occurs after attempted forward. If a user's mailbox was full and + a message was attempted to be forwarded to said box, warnings on + the console would indicate failure. However, the played prompt + was that of success (vm-msgsaved). Now storage failure is taken + into account and the correct prompt (vm-mailboxfull) is played + when appropriate. ABE-2123 SWP-1262 ........ ................ + +2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20 + Apr 2010) | 7 lines Add missing 'useragent' field to + sip-friends.sql file. (closes issue #17171) Reported by: thehar + Patches: sip-friends.patch uploaded by thehar (license 831) + Tested by: pabelanger, thehar ........ + +2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com> + + * main/indications.c, /: Merged revisions 257949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 | + qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line + Change log message to match severity. ........ + + * main/indications.c, /: Merged revisions 257947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 | + qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines + Don't consider a missing indications.conf to be a critical error. + There were many changes in revision 176627 which would avoid the + error that a missing config would have caused. Other than this, + there are no other config files (including asterisk.conf, + surprisingly) that are required. ........ + +2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 257810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 | + twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines + Fix incomplete CDR merge from r195881 Because res/res_features.c + was removed and main/cdr.c added, these changes didn't make it to + trunk and the 1.6.x branches ........ + +2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com> + + * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 + Apr 2010) | 2 lines Removing unused configuration parameters + ........ + +2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 + (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) + | 21 lines Make the mixmonitor thread process audio frames faster + Mantis issue 17078 reports MixMonitor recordings have shorter + durations than the call duration. This was because the mixmonitor + thread was not processing frames from the audiohook fast enough. + The mixmonitor thread would slowly fall behind the most recent + audio frame and when the channel hangs up, the mixmonitor thread + would exit without processing the same number of frames as the + channel; leaving the mixmonitor recording shorter than actual + call duration. This revision fixes this issue by moving the + ast_audiohook_trigger_wait() and the subsequent audiohook.status + check into the block where the ast_audiohook_read_frame() + function returns NULL. (closes issue #17078) Reported by: + geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license + 733) Tested by: dhubbard, geoff2010 Review: + https://reviewboard.asterisk.org/r/611/ ........ ................ + +2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/app.h, /, main/app.c: Merged revisions 257560 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 + (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) + | 6 lines Allow application options with arguments to contain + parentheses, through a variety of escaping techniques. Fixes + SWP-1194 (ABE-2143). Review: + https://reviewboard.asterisk.org/r/604/ ........ ................ + + * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) + | 20 lines Merged revisions 257467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) + | 13 lines Don't recreate peer, when responding to a repeated + deregistration attempt. When a reply to a deregistration is lost + in transmit, the client retries the deregistration. Previously, + this would cause a realtime/autocreate peer to be loaded back + into memory, after it had already been correctly purged. Instead, + we just want to resend the reply without loading the peer. + (closes issue #16908) Reported by: kkm Patches: + 20100412__issue16908.diff.txt uploaded by tilghman (license 14) + Tested by: kkm ........ ................ + +2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com> + + * doc/backtrace.txt: Merged revisions 257427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010) + | 21 lines Merged revisions 257426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) + | 13 lines Update backtrace.txt documentation. Update the + backtrace.txt documentation so it conforms to the same layout as + other documents we've been working on recently. Additionally, add + a bunch of new information about gathering backtraces for crashes + and deadlocks, along with ways of verifying your file before + uploading it. Create a couple of one line commands for people to + generate the files we need. (closes issue #17190) Reported by: + lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen + (license 10) Tested by: lmadsen, pabelanger ........ + ................ + + * doc/backtrace.txt: Merged revisions 257343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010) + | 9 lines Merged revisions 257342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) + | 1 line Update address of the bug tracker. ........ + ................ + +2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com> + + * configs/features.conf.sample, /, main/features.c: Merged + revisions 257262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | + tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 + lines Yet another issue where the conversion of the application + delimiter to comma caused an issue. Application arguments within + the feature map could possibly contain a comma, which conflicts + with the syntax of the features.conf configuration file. This + patch allows the argument to be wrapped in parentheses or quoted, + to allow the application arguments to be interpreted as a single + configuration parameter. (closes issue #16646) Reported by: + pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/547/ ........ + +2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | + tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 + lines Also unref the pvt when we delete the provisional keepalive + job. (closes issue #16774) Reported by: kowalma Patches: + 20100315__issue16774.diff.txt uploaded by tilghman (license 14) + Tested by: falves11, jamicque Review: + https://reviewboard.asterisk.org/r/591/ ........ + +2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com> + + * main/manager.c, /, configs/manager.conf.sample: Merged revisions + 257146 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr + 2010) | 16 lines Merged revisions 257070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr + 2010) | 9 lines Add an option to restore past broken behavor of + the Events manager action Before r238915, certain values for the + EventMask parameter of the Events action would result in no + response being returned. This patch adds an option to restore + that broken behavior. Also while fixing this bug I discovered + that passing an empty EventMasks parameter would also result in + no response being returned, this has been fixed as well while + being preserved when the broken behavior is requested. (closes + issue #17023) Reported by: nblasgen Review: + https://reviewboard.asterisk.org/r/602/ ........ ................ + +2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010) + | 8 lines Ensure that we can have commas within cdr values. + (closes issue #17001) Reported by: snuffy Patches: + 20100412__issue17001.diff.txt uploaded by tilghman (license 14) + Tested by: snuffy ........ + +2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com> + + * doc/HOWTO_collect_debug_information.txt (added): Merged revisions + 256901 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010) + | 23 lines Merged revisions 256900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) + | 15 lines Add How-To document on collecting debugging info for + issues.asterisk.org Paul Belanger has been helping a lot with bug + tracking recently and created this document that we can now point + to when additional debugging information is required. This + document will help those filing issues to know how to get the + information required when filing their issues. This will make + things easier on the developers. Initial text and changes by + pabelanger. Tweaks and editing by myself. (closes issue #17159) + Reported by: pabelanger Patches: + HOWTO_collect_debug_information.txt.patch uploaded by lmadsen + (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ + ................ + + * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010) + | 3 lines Remove silly debug message that is not useful. (issue + #17159) ........ + + * /, main/logger.c: Merged revisions 256821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 | + lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines + CLI command logger set level auto complete. A simple patch to + enable auto tab complete. (closes issue #17152) Reported by: + pabelanger Patches: 0017152.patch uploaded by pabelanger (license + 224) ........ + +2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com> + + * main/app.c: Backport /proc/%d/fd method of closing file + descriptors to 1.6.2. + +2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Merged revisions 256370 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) + | 2 lines Mac OS X does not support comparing a mutex to its + initializer. Create a test for this. ........ + +2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty + after the first call. The bug is exposed if MFC/R2 support is + built into asterisk (i.e., openr2.h is present in the include + path). Code that unconditionally clears the CallerID name and + number is included. Also fixed a malformed if test in mkintf() + added by issue 15883. Converted the if statement to a switch + statement for clarity. Regression of the issue 15883 fix. (closes + issue #16968) Reported by: grecco Patches: issue16968.patch + uploaded by rmudgett (license 664) (closes issue #16747) Reported + by: viniciusfontes + + * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 + (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) + | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by + PRI lock. SWP-1231 ABE-2163 ........ ................ + +2010-05-03 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.7 Released + +2010-04-29 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.7-rc3 Released + +2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com> + + * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in + audiohook_write_list. (closes issue 0017052) Reported by: dvossel + Tested by: dvossel. (closes issue 0016196) Reported by: atis. + Review: https://reviewboard.asterisk.org/r/623/ + +2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com> + + * channels/chan_local.c, main/channel.c: Resolves deadlocks in + chan_local. (closes issue 0017185) Reported by: schmoozecom + Patches: issue_17185_v1.diff uploaded by dvossel (license 671) + issue_17185_v2.diff uploaded by dvossel (license 671) Tested + by: schmoozecom, GameGamer43 + Review: https://reviewboard.asterisk.org/r/631/ + +2010-04-13 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.7-rc2 Released + +2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com> + + Also unref the pvt when we delete the provisional keepalive job. + + (closes issue #16774) + Reported by: kowalma + Patches: + 20100315__issue16774.diff.txt uploaded by tilghman (license 14) + Tested by: falves11, jamicque + + Review: https://reviewboard.asterisk.org/r/591/ + +2010-04-05 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.7-rc1 Released + +2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com> + + * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) + | 1 line Fix for localchannel.tex to allow PDFs to be generated + again. ........ + +2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 | + russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines + Export MEETMEBOOKID and fix pin-less conferences with realtime + conferences (closes issue #16866) Reported by: DEA Patches: + rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA + Review: https://reviewboard.asterisk.org/r/582/ ........ + + * channels/chan_local.c, /: Merged revisions 256015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r256015 | russell | 2010-04-02 18:46:45 -0500 + (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) + | 9 lines Resolve a deadlock that occurs due to a pointless call + to ast_bridged_channel() (closes issue #16840) Reported by: + bzing2 Patches: patch.txt uploaded by bzing2 (license 902) + issue_16840.rev1.diff uploaded by russell (license 2) Tested by: + bzing2, russell ........ ................ + + * main/channel.c, /: Merged revisions 256010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010) + | 9 lines Merged revisions 256009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) + | 2 lines Remove extremely verbose debug message. ........ + ................ + +2010-04-02 20:20 +0000 [r255955] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 255952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | + tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines + Pass the PID of the Asterisk process, not the PID of the canary. + (closes issue #17065) Reported by: globalnetinc Patches: + astcanary.patch uploaded by makoto (license 38) Tested by: frawd, + globalnetinc ........ + +2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) + | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue + #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt + uploaded by tilghman (license 14) ........ + + * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500 + (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) + | 15 lines Ensure line terminators in email are consistent. Fixes + an issue with certain Mail Transport Agents, where attachments + are not interpreted correctly. (closes issue #16557) Reported by: + jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by + tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt + uploaded by tilghman (license 14) + 20100308__issue16557__trunk.diff.txt uploaded by tilghman + (license 14) Tested by: ebroad, zktech Reviewboard: + https://reviewboard.asterisk.org/r/544/ ........ ................ + +2010-03-31 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com> + + * configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 + Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' + can be used. (closes issue #17021) Reported by: kovzol Tested by: + lmadsen, kovzol, davidw, ebroad ........ + +2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com> + + * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r255410 | russell | 2010-03-30 15:56:26 -0500 + (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 + Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does + not start. ........ ................ + + * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010) + | 9 lines Merged revisions 255322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) + | 2 lines Don't make Asterisk not start if pbx_dundi fails to + initialize. ........ ................ + +2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com> + + * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010) + | 6 lines Replace some documentation from 1.6.x back into trunk. + This documentation associated wth tlsbindaddr is still useful so + lets synchronize it between trunk and 1.6.x branches. (issue + #17054) ........ + + * configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) + | 8 lines Update confusing documentation for tlsbindaddr. Update + some confusing documentation for the tlsbindaddr option in + sip.conf.sample. Point at a link instead which has better + documentation. (closes issue #17054) Reported by: klaus3000 + ........ + +2010-03-25 20:43 +0000 [r254770-254805] Jason Parker <jparker@digium.com> + + * utils/Makefile, /: Merged revisions 254802 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) | + 9 lines Merged revisions 254800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | + 1 line Don't remove local copies of utils in uninstall. ........ + ................ + + * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS + issue with out-of-tree modules. Take 2, without ABI breakage this + time. Review: https://reviewboard.asterisk.org/r/588/ + +2010-03-25 20:09 +0000 [r254721] Russell Bryant <russell@digium.com> + + * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) + | 2 lines chan_usbradio depends on alsa. ........ + +2010-03-25 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 + (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar + 2010) | 5 lines Add doxygen for acl.h Review: + https://reviewboard.asterisk.org/r/528 ........ ................ + +2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com> + + * channels/chan_sip.c: Initialize stream to avoid a compilation + error. + +2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix potential crashes from trying to + reference nonexistent RTP streams. + +2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com> + + * /, main/file.c: Merged revisions 254453 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010) + | 9 lines Merged revisions 254451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) + | 2 lines Handle new SRCCHANGE control message here too ........ + ................ + +2010-03-25 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500 + (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar + 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. + Here is a copy and paste of the details from my request on + reviewboard that dealt with these changes: Fix 1. The first + change in place is to fix Mantis issue 15811, which deals with a + situation where Asterisk will incorrectly interpret out of order + RFC2833 frames as duplicate DTMF digits. For instance, we would + receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: + DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 + seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch + when we received the frame with seqno 5, we would interpret this + as a new DTMF 1. With this patch, we will check the seqno of the + incoming digit and not process the frame if the seqno is lower + than the last recorded seqno. Note that we do not record the + seqno of the dropped DTMF frame for future processing. While the + above situation is what was designed to be fixed, the patch is + written in such a way that the following would also be fixed too: + seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) + seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno + 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In + this second situation, the beginning of the DTMF 2 arrives before + the final end frame of the DTMF 1. With the patch, seqno 12 is no + processed and thus we properly interpret the DTMF. Fix 2. The + second change in place is to fix an issue like the following: + seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet + lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) + *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had + code in place that was supposed to properly end the previously + unended DTMF 1. The problem was that the code was essentially a + no-op. The code would set up an end frame for the DTMF 1 but + would immediately overwrite the frame with the begin for DTMF 2. + I changed process_dtmf_rfc2833() so that instead of returning a + single frame, it is given as an output parameter a list of + frames. Each frame that needs to be returned is appended to this + list. Fix 3. The final change is a minor one where an + AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco + DTMF or an RFC 3389 frame and no frame was returned, then we + would return &ast_null_frame. The problem is that earlier in the + function, we may have generated an AST_CONTROL_SRCCHANGE frame + and put it in the list of frames we wish to return. This frame + would be lost in such a case. The patch fixes this problem + ........ ................ + +2010-03-25 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com> + + * /, res/res_agi.c: Merged revisions 254446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 | + lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines + handle_speechset has 4 arguments. Update code to reflect that + handle_speechset has 4 arguments. (closes issue #17093) Reported + by: gpatri Patches: res_agi.patch uploaded by gpatri (license + 1014) Tested by: pabelanger, mmichelson ........ + +2010-03-24 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) + | 78 lines Merged revisions 254235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) + | 72 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248860. As such + the dialplan test has been extended: ; non absolute path, not + combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) + exten => 5040, n, dial(sip/5001) ; absolute path, not combined + exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => + 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, + monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; + combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) + exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, + dial(sip/5001) ; combined: changemonitor from no path to non + absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) + exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this + wasn't possible before exten => 5044, n, dial(sip/5001) ; non + absolute path, combined exten => 5045, 1, + monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, + dial(sip/5001) ; absolute path, combined exten => 5046, 1, + monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, + dial(sip/5001) ; no path, combined exten => 5047, 1, + monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; + combined: changemonitor from non absolute to absolute (leaves + tmp/jeff) exten => 5048, 1, + monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, + changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, + dial(sip/5001) ; combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5049, 1, + monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, + changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, + dial(sip/5001) ; combined: changemonitor from no path to absolute + exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, + changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, + dial(sip/5001) ; combined: changemonitor from absolute to no path + (leaves /tmp/jeff) exten => 5051, 1, + monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, + changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; + not combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) + exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, + dial(sip/5001) ; not combined: changemonitor from no path to non + absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => + 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, + dial(sip/5001) ; not combined: changemonitor from non absolute to + absolute (leaves tmp/jeff) exten => 5054, 1, + monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, + changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, + dial(sip/5001) ; not combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5055, 1, + monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, + changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, + dial(sip/5001) ; not combined: changemonitor from no path to + absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => + 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, + n, dial(sip/5001) ; not combined: changemonitor from absolute to + no path (leaves /tmp/jeff) exten => 5057, 1, + monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, + changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) + ........ ................ + +2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * tests/Makefile, /: Merged revisions 254001 via svnmerge from + http://svn.digium.com/svn/asterisk/trunk ........ r254001 | + tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines + Change the name of the category 'TEST' to match the name of the + subdir ........ + +2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 254050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 | + jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines + Exit native bridging early for greater timing accuracy with + warnings This changes native bridging to break one millisecond + early so that the more accurate timeval calculations done in the + generic bridge can be performed using the bridge config. + Currently the time between exiting native bridging slightly late + can sometimes cause a large enough discrepancy for warnings to be + missed. For the record, 1.4 does not attempt to native bridge at + all when warnings are enabled. (closes issue #15815) Reported by: + adomjan Review: https://reviewboard.asterisk.org/r/577/ ........ + +2010-03-22 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 253800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar + 2010) | 11 lines Merged revisions 253799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar + 2010) | 4 lines Unconditionally copy the caller's account code to + the called party. (related to issue #16331) ........ + ................ + +2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22 + Mar 2010) | 2 lines Update query should be an UPDATE, not a + SELECT. ........ + + * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22 + Mar 2010) | 4 lines Return the list for later manipulation. This + fixes an issue with the update procedure. Debugging with + mmichelson. ........ + + * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged + revisions 253712 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 | + tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines + Accomodate equal signs in DSNs and add documentation, based upon + mmichelson's feedback. ........ + +2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com> + + * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /, + main/features.c: Merged revisions 253540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 | + russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines + Resolve more compiler warnings on FreeBSD. ........ + + * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 + Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD. + ........ + + * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 | + russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines + Resolve a compiler warning on FreeBSD. ........ + + * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) + | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix + build issues I had with this module on FreeBSD. ........ + +2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz> + + * main/astobj2.c, /: Merged revisions 253490 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 | + alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19 + lines prevent segfault if bad magic number is encountered. + internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic + number', but internal_ao2_ref continues on, causing segfault. + Although AO2_MAGIC number is checked by INTERNAL_OBJ before + internal_ao2_ref is called, A02_MAGIC is being destroyed (or a + wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ. + internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad + magic number. (issue #17037) Reported by: alecdavis Patches: + bug17037.diff.txt uploaded by alecdavis (license 585) Tested by: + alecdavis ........ + +2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com> + + * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010) + | 7 lines Change usage of pipe to comma in UserEvent docs. Change + the example usage of pipe as a separator to comma in the + UserEvent documentation. (closes issue #16961) Reported by: + jlpedrosa ........ + + * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010) + | 9 lines Update to new Local channel documentation. Add same + changes as commit to 1.4, but convert to TeX. (issue #16963) + Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz + (license 834) ........ + +2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com> + + * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_mgcp.c, channels/chan_sip.c, + include/asterisk/rtp.h: Revert API change in release branches + This re-renames ast_rtp_update_source to ast_rtp_new_source + +2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com> + + * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 | + lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line + Fix a typo. ........ + + * configs/say.conf.sample, /: Merged revisions 253028 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 + (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) + | 6 lines Add french snipset to say.conf. Add the french snipset + to say.conf. (Closes issue #15799) ........ ................ + +2010-03-16 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 | + tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines + Mask out previous arguments on each nested invocation of Gosub. + (closes issue #16758) Reported by: wdoekes Patches: + 20100316__issue16758.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/561/ ........ + +2010-03-16 19:38 +0000 [r252850] Sean Bright <sean@malleable.com> + + * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar + 2010) | 10 lines Include an extra newline after "Aliased CLI + command" to get back the prompt. The other issue mentioned in + this bug will be more difficult to resolve since we have no idea + (right now) of knowing if the command that is aliased has been + installed yet. (issue #16978) Reported by: jw-asterisk Tested by: + seanbright ........ + +2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com> + + * utils/Makefile, /: Merged revisions 252767 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) + | 13 lines Merged revisions 252766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) + | 6 lines Don't treat warnings as errors for muted. muted + supports OS X, but uses functions marked as deprecated in 10.6. + However, the functions are still supported, so just ignore the + warnings for now and allow the build to proceed. ........ + ................ + +2010-03-16 18:49 +0000 [r252763] Leif Madsen <lmadsen@digium.com> + + * configs/extensions.ael.sample, /: Merged revisions 252762 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 + (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) + | 7 lines Additional extensions.ael global variable fixes. Fixing + up a couple more overlapping global variable namespaces shared + with extensions.conf.sample. Also noticed a few of the lines that + were commented out didn't have the closing semi-colon so I added + that as well. (issue #17035) ........ ................ + +2010-03-15 21:59 +0000 [r252626] Sean Bright <sean@malleable.com> + + * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 | + seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4 + lines Resolve a crash in SLATrunk when the specified trunk + doesn't exist. Reported by philipp64 in #asterisk-dev. ........ + +2010-03-15 21:54 +0000 [r252622] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions + 252619 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010) + | 9 lines Merged revisions 252617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) + | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................ + +2010-03-15 20:53 +0000 [r252535] Leif Madsen <lmadsen@digium.com> + + * configs/extensions.ael.sample: Merged revisions 252534 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 + (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) + | 7 lines Update extensions.ael file to not overlap + extensions.conf. Updated the extensions.ael file so the global + variables don't overlap those that we have in extensions.conf + (sample files). This way unexpected things won't happed hopefully + if both pbx_ael and res_config are loaded. (closes issue #17035) + Reported by: pprindeville ........ ................ + +2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 | + tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines + THIS IS NOT PYTHON. Indentation doesn't matter, only braces do. + (closes issue #17025) Reported by: smurfix Patches: sip.patch + uploaded by smurfix (license 547) ........ + + * main/asterisk.c, Makefile, + contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged + revisions 252362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) + | 11 lines Merged revisions 252361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) + | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: + https://reviewboard.asterisk.org/r/551/ ........ ................ + +2010-03-14 17:48 +0000 [r252317] Sean Bright <sean@malleable.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar + 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They + added a sqlite3_log() function which was conflicting with our + function names. (closes issue #17017) Reported by: alephlg + ........ + +2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com> + + * main/rtp.c: Remove unusued field + + * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, + channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c, + channels/chan_skinny.c, include/asterisk/rtp.h, + channels/chan_h323.c: Merged revisions 252089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | + twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines + Only change the RTP ssrc when we see that it has changed This + change basically reverts the change reviewed in + https://reviewboard.asterisk.org/r/374/ and instead limits the + updating of the RTP synchronization source to only those times + when we detect that the other side of the conversation has + changed the ssrc. The problem is that SRCUPDATE control frames + are sent many times where we don't want a new ssrc, including + whenever Asterisk has to send DTMF in a normal bridge. This is + also not the first time that this mistake has been made. The + initial implementation of the ast_rtp_new_source function also + changed the ssrc--and then it was removed because of this same + issue. Then, we put it back in again to fix a different issue. + This patch attempts to only change the ssrc when we see that the + other side of the conversation has changed the ssrc. It also + renames some functions to make their purpose more clear. Review: + https://reviewboard.asterisk.org/r/540/ ........ + +2010-03-12 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1 + line add missing mfcr2_skip_category setting ........ + +2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010) + | 8 lines Don't override a user option with the global option. + (closes issue #16849) Reported by: ip-rob Patches: + 20100311__issue16849.diff.txt uploaded by tilghman (license 14) + Tested by: ip-rob ........ + +2010-03-12 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010) + | 1 line Doxegen this chan_dahdi lock. ........ + +2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com> + + * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 | + tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines + Because ExecIf needs to reprocess arguments, it's best if we + don't remove quotes during parsing. (closes issue #16905) + Reported by: ip-rob Patches: 20100303__issue16905.diff.txt + uploaded by tilghman (license 14) Tested by: ip-rob ........ + + * apps/app_system.c, /: Merged revisions 251877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 | + tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines + If the argument to the system application is quoted, ensure we + remove the quotes before trying to execute. (closes issue #16842) + Reported by: ip-rob Patches: 20100310__issue16842.diff.txt + uploaded by tilghman (license 14) Tested by: ip-rob ........ + +2010-03-11 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.6 released + +2010-03-05 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.6-rc2 released + +2010-03-05 Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman + | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in + ODBC query. (closes issue #16953) Reported by: elguero Patches: + app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) + ........ + +2010-03-04 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.2.6-rc1 released + +2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com> + + * doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) + | 11 lines Update existing Local channel documentation. A + complete re-write of the Local channel documentation has been + performed, with the existing information from localchannel.txt + and localchannel.tex merged in. (closes issue #16637) Reported + by: kobaz Patches: localchannel.tex uploaded by lmadsen (license + 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: + lmadsen, jsmith, mmichelson ........ + +2010-03-03 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 + (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) + | 15 lines Make sure to clear red alarm after polarity reversal. + From the issue: The automatic overnight line tests (or manual + ones) used on UK (BT) lines causes a red alarm on a dahdi / + TDM400P connected channel. This is because the line uses voltage + tests (battery loss) and polarity reversal. The polarity reversal + causes chan_dahdi to initiate v23 CallerID processing but during + this the event DAHDI_EVENT_NOALARM is ignored so that the alarm + is never cleared. (closes issue #14163) Reported by: jedi98 + Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license + 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ + ................ + +2010-03-03 18:05 +0000 [r250253-250396] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 + (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) + | 16 lines fixes problem with duplicate TXREQ packets When + Asterisk receives an IAX2 TXREQ packet, try_transfer() will call + store_by_transfercallno() to link the chan_iax2_pvt struct into + iax_transfercallno_pvts. If a duplicate TXREQ packet is received + for the same call, the pvt struct will be linked into + iax_transfercallno_pvts multiple times. This patch fixes this. + Thanks rain for debugging this and providing a patch! (closes + issue #16904) Reported by: rain Patches: + iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested + by: rain, dvossel ........ ................ + + * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | + dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines + fixes signed to unsigned int comparision issue for FaxMaxDatagram + value. ........ + +2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com> + + * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) + | 8 lines Update IMAP documentation. Update the IMAP + documentation to make it clear that storing voicemails in the + same folder as a large number of emails could potentially cause + significant slow downs when writing or retrieving voicemails. + (issue #16704) Reported by: TimeHider Tested by: lmadsen, + TimeHider ........ + + * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 + (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) + | 7 lines Update documentation to clarify purpose of unanswered + option. (closes issue #16267) Reported by: elsto Patches: + cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested + by: davidw, elsto ........ ................ + + * doc/tex/configuration.tex, /: Merged revisions 250037 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 + Mar 2010) | 4 lines Update documentation to not imply we support + overriding options. (closes issue #16855) Reported by: davidw + ........ + + * apps/app_directory.c, /: Merged revisions 249950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) + | 4 lines Fix literal values wrapped in documentation. (closes + issue #16145) Reported by: tilghman ........ + +2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz> + + * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c, + UPGRADE.txt: revert ability to exit echo app caused a regression, + as only supported VOICE, not VIDEO etc. (issue #16880) + +2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com> + + * /, main/features.c: Merged revisions 249925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 | + lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines + Add missing description of the PARKINGLOT variable in XML + documentation. (closes issue #16743) Reported by: snuffy Patches: + parkingdoc.diff uploaded by snuffy (license 35) ........ + + * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 | + lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines + Convert some DUNDI functions to XML documentation. (closes issue + #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by + snuffy (license 35) ........ + +2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com> + + * channels/chan_console.c, channels/chan_gtalk.c, + channels/chan_oss.c, channels/misdn_config.c, + include/asterisk/abstract_jb.h, configs/alsa.conf.sample, + channels/chan_jingle.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, channels/chan_skinny.c, + configs/mgcp.conf.sample, main/abstract_jb.c, + channels/chan_h323.c, channels/chan_alsa.c, + configs/sip.conf.sample, channels/chan_mgcp.c, + channels/chan_unistim.c, configs/console.conf.sample, + configs/chan_dahdi.conf.sample, channels/chan_local.c, + configs/oss.conf.sample, channels/chan_sip.c, /, + configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged + revisions 249893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | + dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines + fixes adaptive jitterbuffer configuration When configuring the + adaptive jitterbuffer, the target_extra value not only could not + be set from the configuration, but was not even being set to its + proper default. This value is required in order for the adaptive + jitterbuffer to work correctly. To resolve this a config option + has been added to expose this value to the conf files, and a + default value is provided when no config specific value is + present. ........ + +2010-03-02 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com> + + * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) + | 1 line Fix several XML documentation validate errors. ........ + +2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz> + + * apps/app_echo.c: fixes ability to exit echo app when called from + a ISDN channel, null frames prevent '#' exit. Now only echo back + VOICE and DTMF frames (issue #16880) Reported by: alecdavis + Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis + +2010-03-01 19:40 +0000 [r249675] Sean Bright <sean@malleable.com> + + * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 + (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar + 2010) | 11 lines Fix crash in app_voicemail related to message + counting. We were passing a 'struct inprocess **' and treating it + like a 'struct inprocess *' causing a segfault. (closes issue + #16921) Reported by: whardier Patches: 20100301_issue16921.patch + uploaded by seanbright (license 71) Tested by: whardier ........ + ................ + +2010-03-01 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) + | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP + compile once again. ........ + +2010-03-01 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 249538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 + (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) + | 11 lines Modify queued frames from local channels to not set + the other side to up In this case, attended transfers were broken + due to ast_feature_request_and_dial detecting the channel being + set to up before the answer frame could be read and therefore + failing to mark the channel as ready. This fix is a regression + fix for 244785, which should continue to work properly as well. + (closes issue #16816) Reported by: jamhed Tested by: jamhed, + corruptor ........ ................ + +2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) + | 5 lines Fix unit test that Alec Davis broke. (closes issue + #16927) Reported by: alecdavis ........ + + * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions + 249405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | + tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines + Properly document voicemail API documents. Also fix a crash + reported via the -dev list. ........ + +2010-02-27 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz> + + * channels/chan_dahdi.c: overlap receiving: automatically send CALL + PROCEEDING when dialplan starts Following Q.931 5.2.4 When the + user has determined that sufficient call information has been + received the user shall stop T302 and send CALL PROCEEDING to the + network. Previously timeouts were possible if the dialplan took a + long time to issue any response back to the network. Verified + that our local TELCO also does the same. (issue #16789) Reported + by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded + by alecdavis (license 585) Tested by: alecdavis + +2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 + (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 + Feb 2010) | 1 line add a reference to the now-published IAX2 RFC + ........ ................ + +2010-02-26 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) + | 18 lines Cleanups to fix bugs in the VM count API functions. - + Urgent voicemails were not attached, because the attachment code + looked in the wrong folder. - Urgent voicemails were sometimes + counted twice when displaying the count of new messages. - + Backends were inconsistent as to which voicemails each API + counted. (closes issue #15654) Reported by: tomo1657 Patches: + 20100225__issue15654.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman (closes issue #16448) Reported by: hevad + Review: https://reviewboard.asterisk.org/r/525/ ........ + +2010-02-26 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb + 2010) | 14 lines Merged revisions 249100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb + 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. + (closes issue #16792) Reported by: vrban Patches: t38_606.patch + uploaded by vrban (license 756) ........ ................ + +2010-02-25 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) + | 24 lines Merged revisions 248860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) + | 18 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248757. As such + the dialplan test has been extended: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) exten => 5043, 1, + monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, + changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) + exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, + changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by + design and emits a warning exten => 5044, n, dial(sip/5001) + ........ ................ + +2010-02-25 22:42 +0000 [r248949] Mark Michelson <mmichelson@digium.com> + + * /, main/acl.c: Merged revisions 248946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | + mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 + lines Fix incorrect ACL behavior when CIDR notation of "/0" is + used. AST-2010-003 ........ + +2010-02-25 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 248861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) + | 22 lines Merged revisions 248859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) + | 15 lines Some platforms clear /var/run at boot, which makes + connecting a remote console... difficult. Previously, we only + created the default /var/run/asterisk directory at install time. + While we could create it in the init script, that would not work + for those who start asterisk manually from the command line. So + the safest thing to do is to create it as part of the Asterisk + boot process. This also changes the ownership of the directory, + because the pid and ctl files are created after we setuid/setgid. + (closes issue #16802) Reported by: Brian Patches: + 20100224__issue16802.diff.txt uploaded by tilghman (license 14) + Tested by: tzafrir ........ ................ + +2010-02-25 18:52 +0000 [r248797] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) + | 22 lines Merged revisions 248757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) + | 15 lines Ensure that monitor recordings are written to the + correct location. Recordings should be placed in the monitor + directory when a non-absolute path is used. Exact dialplan used + for testing: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) ABE-2101 ........ + ................ + +2010-02-24 21:29 +0000 [r248643] Tilghman Lesher <tlesher@digium.com> + + * /, main/logger.c: Merged revisions 248584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) + | 14 lines Merged revisions 248582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) + | 7 lines Remove color code sequences from verbose messages that + go to logfiles. (closes issue #16786) Reported by: dodo Patches: + logger2.patch uploaded by dodo (license 989) Tested by: tilghman + ........ ................ + +2010-02-23 16:37 +0000 [r248398] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) + | 15 lines Merged revisions 248396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) + | 9 lines fixes invite with replaces deadlock (closes issue + #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 + uploaded by dvossel (license 671) Tested by: pwalker, dvossel + ........ ................ + +2010-02-19 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_console.c, main/loader.c, /: Merged revisions + 228798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 | + tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 + lines Fix various problems detected with Valgrind. * chan_console + accessed pvts after deallocation. * The module loader did not + check usecount on shutdown, which led to chan_iax2 reading a + timer that was already unloaded. (closes issue #16062) Reported + by: alexanderheinz Patches: 20091109__issue16062.diff.txt + uploaded by tilghman (license 14) Tested by: tilghman ........ + +2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 + line mfcr2 issue 0016844 - Fix portability bit fields and make + mfcr2_immediate_accept work again, reported and patched by + korihor ........ + +2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 + (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 + (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... + .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, + 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing + consistent with other channel technologies. The processing of + DTMF tones on the receiving side of an ISDN channel is + inconsistent with the way it is handled in other channels, + especially DAHDI analog. This causes DTMF tones sent from an ISDN + phone to be doubled at the connected party. We are using the + following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes + Option one is necessary because the asterisk DSP DTMF detection + is better than mISDN's internal DSP. Not as many false positives. + Option two is necessary to transmit DTMF tones end to end when + mISDN channels are connected to SIP channels with out of band + DTMF for example. The symptom is that DTMF tones sent by an ISDN + phone are doubled on the way through asterisk when two mISDN + channels are connected with a Local channel in between or if it + is bridged to an analog channel. The doubling of DTMF tones is + because DTMF is passed inband to asterisk by the mISDN channel + and passed out of band once again after the release of the DTMF + tone. Passing it inband is wrong. Neither an analog channel nor + SIP channel passes DTMF inband if configured to inband DTMF. + Analog and SIP channels filter out the DTMF tones because they + use the voice frames returned by ast_dsp_process. But chan_misdn + passes the unfiltered input voice frames instead. To overcome one + aspect of the problem, the doubling of DTMF tones when two mISDN + channels are directly bridged, someone made an 'optimization', + where in that case the DTMF tone passed out-of-band to the peer + channel is not translated to an inband tone at the transmit side. + This optimization is bad because it does not work in general. For + example, analog channels or mISDN channels when bridged through + an intermediary local channel will generate DTMF tones from + out-of-band information. Also, of course, it must not be done + when there is no inband DTMF available. This patch fixes the + issue. Now chan_misdn will filter the received inband DTMF signal + the same as other channel types. Another change included: No need + to build an extra translation path because ast_process_dsp does + it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 + ................ ................ + +2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | + dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines + handle_request_invite revise comment, fix coding guideline issues + I'm working with this code right now trying to analyze a + deadlock. This change is just to clean up a few things before I + make a more complex patch. ........ + +2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher <tlesher@digium.com> + + * res/res_speech.c, /: Merged revisions 247841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | + tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines + Revert an errant part of a previous cleanup, to fix a memory + corruption issue. (closes issue #16368) Reported by: thirionjwf + Patches: res_speech.c.patch uploaded by thirionjwf (license 955) + ........ + + * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | + tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 + lines If the peer record is from realtime, it could be set to 0, + due to MySQL not representing NULL well in integer columns. NULL + means the value is not specified for the column, which normally + means the driver uses whatever is the default value. However, on + MySQL, placing a NULL in either a float or integer column results + in a retrieval of the 0 value. Hence, users get an errant error + on load. This patch suppresses that error and makes the value as + if it was not there. Note that this cannot be done in the + realtime driver, because the lack of difference between NULL and + 0 can only be intepreted correctly by the driver itself. If we + did it in the realtime driver, then it would be effectively + impossible to set any realtime field to 0, because it would act + as if the field were unspecified and possibly take on a different + value. (closes issue #16683) Reported by: wdoekes ........ + +2010-02-18 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com> + + * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) + | 9 lines fixes confbridge crash when no timing module is loaded. + (closes issue #16471) Reported by: kjotte Patches: M16471.diff + uploaded by junky (license 177) Tested by: kjotte, junky ........ + + * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | + dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines + fixes Queue with C option crash (closes issue #16475) Reported + by: okrief Patches: queue_crash.diff uploaded by dvossel (license + 671) ........ + +2010-02-18 19:41 +0000 [r247653] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 247652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb + 2010) | 13 lines Merged revisions 247651 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb + 2010) | 6 lines Copy the calling party's account code to the + called party if they don't already have one. (closes issue + #16331) Reported by: bluefox Tested by: mnicholson ........ + ................ + +2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen <lmadsen@digium.com> + + * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 + (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 + Feb 2010) | 1 line Add additional link to best practices document + per jsmith. ........ ................ + + * README-SERIOUSLY.bestpractices.txt (added): Merged revisions + 247503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) + | 18 lines Merged revisions 247502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) + | 10 lines Add best practices documentation. (issue #16808) + Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis + Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/507/ ........ ................ + +2010-02-18 04:21 +0000 [r247426] Russell Bryant <russell@digium.com> + + * sounds/Makefile, Makefile, /: Merged revisions 247423 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247423 | russell | 2010-02-17 22:20:11 -0600 + (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) + | 10 lines Tweak argument handling for wget in the sounds + Makefile. 1) Fix the check to see if we are using wget to not be + full of fail. The configure script populates this variable with + the absolute path to wget if it is found, so it didn't work. 2) + Allow some extra arguments to be passed in for wget. This is just + a simple change to allow our Bamboo build script to tell wget to + be quiet and not fill up our logs with download status output. + ........ ................ + +2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/strings.h, main/strings.c, /: Merged revisions + 247335 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | + mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 + lines Fix two problems in ast_str functions found while writing a + unit test. 1. The documentation for ast_str_set and + ast_str_append state that the max_len parameter may be -1 in + order to limit the size of the ast_str to its current allocated + size. The problem was that the max_len parameter in all cases was + a size_t, which is unsigned. Thus a -1 was interpreted as + UINT_MAX instead of -1. Changing the max_len parameter to be + ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an + off-by-one error in the case where we attempted to write a string + larger than the current allotted size to a string when -1 was + passed as the max_len parameter. When trying to write more than + the allotted size, the ast_str's __AST_STR_USED was set to 1 + higher than it should have been. Thanks to Tilghman for quickly + spotting the offending line of code. Oh, and the unit test that I + referenced in the top line of this commit will be added to + reviewboard shortly. Sit tight... ........ + + * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb + 2010) | 9 lines Merged revisions 247168 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb + 2010) | 3 lines Make sure that when autofill is disabled that + callers not in the front of the queue cannot place calls. + ........ ................ + + * main/strings.c, /: Merged revisions 247076 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 | + mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12 + lines Add va_end calls to __ast_str_helper. According to the man + page for stdarg(3), "Each invocation of va_copy() must be matched + by a corresponding invocation of va_end() in the same function." + There were several cases in __ast_str_helper where va_copy was + not matched with a corresponding call to va_end. ........ + + * include/asterisk/strings.h, /: Merged revisions 246985 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, + 16 Feb 2010) | 3 lines Add some clarifying documentation to the + ast_str_set and ast_str_append functions. ........ + +2010-02-16 21:03 +0000 [r246900-246982] David Vossel <dvossel@digium.com> + + * main/tcptls.c, /: Merged revisions 246980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | + dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines + warning message if openssl support is missing while attempting + tls connection (closes issue #16673) Reported by: michaesc + Patches: tls_error_msg.diff uploaded by dvossel (license 671) + ........ + + * main/channel.c: fixes merge error with Monitor calculation fix + + * main/channel.c, /: Merged revisions 246899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | + dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines + fixes sample rate conversion issue with Monitor application When + using ast_seekstream with the read/write streams of a monitor, + the number of samples we are seeking must be of the same rate as + the stream or the jump calculation will be incorrect. This patch + adds logic to correctly convert the number of samples to jump to + the sample rate the read/write stream is using. For example, if + the call is G722 (16khz) and the read/write stream is recording a + 8khz wav, seeking 320 samples of 16khz audio is not the same as + seeking 320 samples of 8khz audio when performing the + ast_seekstream on the stream. ABE-2044 ........ + +2010-02-15 23:45 +0000 [r246713] Tilghman Lesher <tlesher@digium.com> + + * Makefile, /: Merged revisions 246710 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) + | 12 lines Merged revisions 246709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) + | 5 lines Make the menuselect instructions correct by allowing + 'make menuselect' to actually solve dependency problems. + (Previously, it would fail out again with the same message about + running 'make menuselect', which was NOT at all helpful.) + ........ ................ + +2010-02-12 23:33 +0000 [r246547] David Vossel <dvossel@digium.com> + + * main/channel.c, /: Merged revisions 246546 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) + | 21 lines Merged revisions 246545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) + | 16 lines lock channel during datastore removal On channel + destruction the channel's datastores are removed and destroyed. + Since there are public API calls to find and remove datastores on + a channel, a lock should be held whenever datastores are removed + and destroyed. This resolves a crash caused by a race condition + in app_chanspy.c. (closes issue #16678) Reported by: + tim_ringenbach Patches: datastore_destroy_race.diff uploaded by + tim ringenbach (license 540) Tested by: dvossel ........ + ................ + +2010-02-12 19:08 +0000 [r246464] Jason Parker <jparker@digium.com> + + * main/channel.c: Fix some silly formatting that made my head hurt. + +2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) + | 2 lines Fussy compiler on another machine... ........ + + * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) + | 2 lines Fix weird issue with unit tests on optimized build - + turned out to be a signing issue. ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_agi.c: Merged revisions 246030 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 | + tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 + lines Solaris doesn't like outputting a NULL to a %s in format + strings. Detect all platforms that don't like that, either, and + ensure that when documentation is missing, we pass a non-NULL + pointer when outputting the corresponding documentation. (closes + issue #16689) Reported by: bklang Patches: + 20100209__issue16689__with_tests.diff.txt uploaded by tilghman + (license 14) Review: https://reviewboard.asterisk.org/r/497/ + ........ + +2010-02-10 17:51 +0000 [r246117] David Vossel <dvossel@digium.com> + + * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) + | 14 lines Merged revisions 246115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) + | 8 lines fixes random deadlock in app_queue with use_weight + during reload (closes issue #16677) Reported by: tim_ringenbach + Patches: app_queue_use_weight_deadlock.diff uploaded by tim + ringenbach (license 540) ........ ................ + +2010-02-10 16:58 +0000 [r246073] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 246070 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) + | 22 lines Change channel state on local channels for + busy,answer,ring. Previously local channels channel state never + changed. This became problematic when the state of the other side + of the local channel was lost, for example during a masquerade. + Changing the state of the local channel allows for the scenario + to be detected when the channel state is set to ringing, but the + peer isn't ringing. The specific problem scenario is described in + 164201. Although this was noted on one of the issues, here is the + tested dialplan verified to work: exten => + 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => + *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) + exten => *9700,n,wait(3) ;3 works, 1 did not exten => + *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did + not exten => + 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes + issue #14992) Reported by: davidw ........ + +2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) + | 2 lines Enable warnings on atypical conditions for the FILTER + function (suggested by mmichelson on the -dev list). ........ + + * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged + revisions 245945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) + | 9 lines Merged revisions 245944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) + | 2 lines Include examples of FILTER usage in extension patterns + where a "." may be a risk. ........ ................ + +2010-02-09 23:11 +0000 [r245794] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 + (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) + | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = + 32768 which is the maximum allowed iax2 callnumber. Creating the + iaxs and iaxsl array of size 32768 means the maximum callnumber + is actually out of bounds. This causes a nasty crash. (closes + issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded + by dvossel (license 671) ........ ................ + +2010-02-09 18:09 +0000 [r245732] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | + tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines + Ensure frames are only freed once. (closes issue #16361) Reported + by: vlad Patches: 20100208__issue16361.diff.txt uploaded by + tilghman (license 14) Tested by: kenny, bloodoff, misaksen + ........ + +2010-02-09 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 | + mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2 + lines This commit removes an extra newline in T.38 generated SDP + packets. This bug was caused by the fix introduced in r243860. + (closes issue #16766) Reported by: raivisr Patches: + t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) + Tested by: raivisr ........ + +2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | + kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 + lines Don't offer MMR or JBIG transcoding during T.38 + negotiation. After further discussion with Steve Underwood, we + should not (yet) be offering to receive MMR or JBIG transcoded + streams from T.38 endpoints. A future spandsp release will + support those features, and then they can be enabled during + negotiation ........ + +2010-02-08 23:47 +0000 [r245626] Russell Bryant <russell@digium.com> + + * /, main/event.c: Merged revisions 245624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 | + russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines + Fix return value of get_ie_str() and get_ie_str_hash() for + non-existent IE. I found this bug while developing a unit test + for event allocation. Testing is awesome. ........ + +2010-02-08 22:46 +0000 [r245581] Tilghman Lesher <tlesher@digium.com> + + * channels/Makefile, /, main/Makefile: Merged revisions 245578 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 + Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and + channels/ Makefiles. They were previously passed correctly, but + they simply weren't used. This caused issues with various + platforms whose builds needed to pass special linker flags via + the configure script. (closes issue #16596) Reported by: + pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by + pprindeville (license 347) Tested by: tilghman ........ + +2010-02-08 20:43 +0000 [r245500] Jason Parker <jparker@digium.com> + + * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 + (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | + 4 lines Remove reference of documentation in source directory. + People don't always build Asterisk from source (distro packages, + anybody?). ........ ................ + +2010-02-05 19:27 +0000 [r245097] Jeff Peeler <jpeeler@digium.com> + + * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 + (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb + 2010) | 5 lines Remove contrib/firmware directory as it is empty + Remove explicit license for IAXy firmware as it is no longer + included in the tree ........ ................ + +2010-02-05 17:10 +0000 [r244930] Sean Bright <sean@malleable.com> + + * main/asterisk.c, /: Merged revisions 244927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb + 2010) | 9 lines Merged revisions 244926 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb + 2010) | 1 line Update main copyright date. ........ + ................ + +2010-02-03 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com> + + * main/sched.c, /: Merged revisions 244547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 | + mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3 + lines Initialize counters in ast_sched_report so that resulting + data is not bogus. ........ + +2010-02-03 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions + 244505 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | + tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines + The chanvar= setting should inherit the entire list of variables, + not just the first one. (closes issue #16359) Reported by: raarts + Patches: dahdi-setvars.diff uploaded by raarts (license 937) + Tested by: raarts ........ + +2010-02-02 22:29 +0000 [r244445] David Vossel <dvossel@digium.com> + + * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: + Merged revisions 244443 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | + dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines + fixes crash during T.38 negotiation caused by invalid or missing + FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported + by: krn (closes issue #16724) Reported by: barthpbx (closes issue + #16517) Reported by: bklang (closes issue #16485) Reported by: + elsto ........ + +2010-02-02 20:35 +0000 [r244395] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 | + tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 + lines Properly respect GOSUB_RESULT as to what to do with the + master channel. Previously, we would parse GOSUB_RESULT, but not + actually do anything with it. (closes issue #16686) Reported by: + bklang Patches: app_dial-respect-gosub_result.patch uploaded by + bklang (license 919) (with modifications) ........ + +2010-02-02 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.2 + + * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can + remotely crash Asterisk by modifying the FaxMaxDatagram field of + the SDP to contain either a negative or exceptionally large value. + The same crash occurs when the FaxMaxDatagram field is omitted from + the SDP as well. + +2010-01-14 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.1 + +2010-01-08 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.1-rc1 + +2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_console.c, channels/chan_oss.c, main/poll.c, + channels/chan_usbradio.c, include/asterisk/utils.h, /, + channels/chan_sip.c, channels/chan_alsa.c: Merged revisions + 209400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | + kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 + lines Define side-effect-safe MIN and MAX macros and remove + duplicate definitions from various files. (closes issue #16251) + Reported by: asgaroth ........ + +2010-01-07 20:17 +0000 [r238362-238416] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 + (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) + | 10 lines fixes crash in "scheduled_destroy" in chan_iax A + signed short was used to represent a callnumber. This is makes it + possible to attempt to access the iaxs array with a negative + index. (closes issue #16565) Reported by: jensvb ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | + dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines + Change in sip show channels display format allowing more digits + for CID (closes issue #16459) Reported by: Rzadzins Patches: + chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) + ........ + + * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | + dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines + cli 'queue show' formatting fix. queue name was truncated over 12 + characters (closes issue #16078) Reported by: RoadKill Patches: + quequename_limit.patch uploaded by ppyy (license 906) Tested by: + dvossel ........ + +2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | + 2 lines Document the usefulness of explicit udp:// in the + register string ........ + +2010-01-06 21:48 +0000 [r238234] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) + | 11 lines Merged revisions 238230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) + | 4 lines Revise documentation on disposition values to the + actual values used. (closes issue #16289) Reported by: wdoekes + ........ ................ + +2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | + jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines + Fix misreverting from 177158. (closes issue #15725) Reported by: + shanermn Patches: v1-15725.patch uploaded by dimas (license 88) + Tested by: shanermn ........ + + * /, main/features.c: Merged revisions 238134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | + jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines + Fix channel name comparison for bridge application. The channel + name comparison was not comparing the whole string and therefore + if one channel name was a substring of the other, the bridge + would fail. (closes issue #16528) Reported by: telecos82 Patches: + res_features_r236843.diff uploaded by telecos82 (license 687) + ........ + +2010-01-06 15:22 +0000 [r238013] Russell Bryant <russell@digium.com> + + * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) + | 14 lines Merged revisions 238009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) + | 7 lines Resolve a crash due to an ast_frame not being fully + initialized. (closes issue #16531) Reported by: john8675309 + (closes SWP-615) ........ ................ + +2010-01-06 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 | + tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines + Whoa, duplicate setting (dead code). ........ + +2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com> + + * apps/app_test.c: Add a wait to ensure TestServer thinks it has + finished sending the final digit. This was previously committed + to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and + missed). 1.6.2 also needs this patch to resolve the bug. (closes + issue #16550) Reported by: opticron Patches: apptest.diff + uploaded by opticron (license 267) + +2010-01-05 23:09 +0000 [r237840-237921] David Vossel <dvossel@digium.com> + + * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | + dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines + fixes holdtime playback issue in app_queue When reporting hold + time, the number of seconds should be mod 60. Otherwise audio + playback could be something like "2 minutes 123 seconds" rather + than "2 minutes 3 seconds". Also, the "minute" sound file is + missing, so for the moment until that file can be created the + "minutes" file is used instead. (closes issue #16168) Reported + by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by + nickilo (license ) Tested by: nickilo, wonderg ........ + + * main/pbx.c, /: Merged revisions 237839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | + dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines + fixes subscriptions being lost after 'module reload' During a + module reload if multiple extension configs are present, such as + both extensions.conf and extensions.ael, watchers for one + config's hints will be lost during the merging of the other + config. This happens because hint watchers are only preserved for + the current config being merged. The old context list is + destroyed after the merging takes place, meaning any watchers + that were not perserved will be removed. Now all hints are + preserved during merging regardless of what config file is being + merged. These hints are only restored if they are present within + the new context list. (closes issue #16093) Reported by: jlaroff + ........ + +2010-01-05 17:25 +0000 [r237743] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 237699 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) + | 14 lines Merged revisions 237697 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) + | 7 lines Change a NOTICE log message to DEBUG where it belongs. + (closes issue #16479) Reported by: alexrecarey (closes SWP-577) + ........ ................ + +2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) + | 6 lines Make CLI command 'mixmonitor start|stop <channel> work + again. (closes issue #16534) Reported by: jlaguilar Fix as + suggested by jlaguilar in the bugreport ........ + +2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c: Merged revisions 237574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) + | 13 lines Merged revisions 237573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) + | 6 lines Bounds checking for input string (closes issue #16407) + Reported by: qwell Patches: 20100104__issue16407.diff.txt + uploaded by tilghman (license 14) ........ ................ + + * main/pbx.c, /: Merged revisions 237494 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) + | 15 lines Merged revisions 237493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) + | 8 lines Regression in issue #15421 - Pattern matching (closes + issue #16482) Reported by: wdoekes Patches: + astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) + 20091223__issue16482.diff.txt uploaded by tilghman (license 14) + Tested by: wdoekes, tilghman ........ ................ + + * main/config.c, /: Merged revisions 237414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | + tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines + Oops, didn't compile (thanks, kpfleming) ........ + + * main/config.c, /: Merged revisions 237410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | + tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines + Further reduce the encoded blank values back to blank in the + realtime API. (closes issue #16533) Reported by: sergee Patches: + 200100104__issue16533.diff.txt uploaded by tilghman (license 14) + Tested by: sergee ........ + + * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged + revisions 237406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) + | 23 lines Merged revisions 237405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) + | 16 lines Add a flag to disable the Background behavior, for AGI + users. This is in a section of code that relates to two other + issues, namely issue #14011 and issue #14940), one of which was + the behavior of Background when called with a context argument + that matched the current context. This fix broke FreePBX, + however, in a post-Dial situation. Needless to say, this is an + extremely difficult collision of several different issues. While + the use of an exception flag is ugly, fixing all of the issues + linked is rather difficult (although if someone would like to + propose a better solution, we're happy to entertain that + suggestion). (closes issue #16434) Reported by: rickead2000 + Patches: 20091217__issue16434.diff.txt uploaded by tilghman + (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by + tilghman (license 14) Tested by: rickead2000 ........ + ................ + +2010-01-04 16:50 +0000 [r237328] David Vossel <dvossel@digium.com> + + * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 | + dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines + app_queue segfaults if realtime field uniqueid is NULL (closes + issue #16385) Reported by: haakon Patches: app_queue.c.patch + uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by + dvossel (license 671) Tested by: haakon ........ + +2010-01-04 16:27 +0000 [r237326] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_agi.c: Merged revisions 237323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | + jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines + Fix timeout for AGI command speech recognize. (closes issue + #16297) Reported by: semond ........ + +2010-01-04 16:21 +0000 [r237322] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 237319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 + (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) + | 3 lines It's also possible for the Local channel to directly + execute an Application. Reviewboard: + https://reviewboard.asterisk.org/r/452/ ........ ................ + +2010-01-02 10:03 +0000 [r237139] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 + lines Merged revisions 237135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 + lines Release memory of the contact acl before unloading module + ........ ................ + +2009-12-30 22:00 +0000 [r236985] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 236982 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 + (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) + | 9 lines Don't queue frames to channels that have no means to + process them. (closes issue #15609) Reported by: aragon Patches: + 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by + tilghman (license 14) Tested by: aragon Review: + https://reviewboard.asterisk.org/r/452/ ........ ................ + +2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler <jpeeler@digium.com> + + * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | + jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines + One more LOW_MEMORY compile fix. ........ + + * main/cli.c, /: Merged revisions 236893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 | + jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines + Fix compiling with LOW_MEMORY. Modified handle_verbose to be + LOW_MEMORY aware. (closes issue #16381) Reported by: + michael_iedema Patches: ast_complete_source_filename.patch + uploaded by michael iedema (license 942) modified by me ........ + +2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) + | 4 lines When the field is blank, don't warn about the field + being unable to be coerced, just skip the column. (closes + http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) + Reported by Nic Colledge on the -dev list, fixed by me. ........ + + * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | + tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines + Shut down the SIP session timers more gracefully, in order to + prevent a possible crash. (closes issue #16452) Reported by: + corruptor Patches: 20091221__issue16452.diff.txt uploaded by + tilghman (license 14) Tested by: corruptor ........ + +2009-12-28 22:13 +0000 [r236716] Jason Parker <jparker@digium.com> + + * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec + 2009) | 8 lines Allow "REMAINDER" to function properly in + expressions. (closes issue #16427) Reported by: wdoekes Patches: + ast16-reminder-remainder.patch uploaded by wdoekes (license 717) + Tested by: wdoekes ........ + +2009-12-28 17:40 +0000 [r236670] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) + | 4 lines Use recommended option, not deprecated option. (closes + issue #16515) Reported by: ManChicken ........ + +2009-12-28 15:31 +0000 [r236513-236635] Sean Bright <sean@malleable.com> + + * include/asterisk/threadstorage.h, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 236613 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec + 2009) | 14 lines Merged revisions 236585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec + 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT + requires extra braces. There was conditional code (based on build + platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that + was removed since it is fixed in newer versions of + Solaris/OpenSolaris, but I am still running into it on Solaris 10 + x86 so add a configure-time check for it. ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec + 2009) | 19 lines Merged revisions 236509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec + 2009) | 12 lines Avoid a crash with large numbers of MeetMe + conferences. Similar to changes made to Queue(), when we have + large numbers of conferences in meetme.conf (1000s) and we use + alloca()/strdupa(), we can blow out the stack and crash, so + instead just use a single fixed buffer. (closes issue #16509) + Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded + by seanbright (license 71) Tested by: seanbright ........ + ................ + +2009-12-27 18:22 +0000 [r236437] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 + (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 + Dec 2009) | 2 lines Turn on colors in the daemon, since there's + many requests for it on Ubuntu. ........ ................ + +2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile, /: Merged revisions 236358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec + 2009) | 9 lines Merged revisions 236357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec + 2009) | 1 line update to latest releases with zero uid/gid + ........ ................ + +2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | + tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines + AGI may be invoked from outside the dialplan (closes issue + #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt + uploaded by tilghman (license 14) Tested by: atis ........ + + * /, res/res_agi.c: Merged revisions 236186 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) + | 11 lines Merged revisions 236184 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) + | 4 lines If EXEC only gets a single argument, don't crash when + the second is used. (closes issue #16504) Reported by: bklang + ........ ................ + +2009-12-22 17:04 +0000 [r236064] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) + | 18 lines Merged revisions 236062 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) + | 11 lines fixes issue with p->method incorrectly set to ACK It + is possible for a second ACK to come in for a retransmitted + message. If an ack does not match an unacked message in our + queue, restore the previous p->method as this ACK is completely + ignored. (closes issue #16295) Reported by: omolenkamp Patches: + issue16295_v2.diff uploaded by dvossel (license 671) ........ + ................ + +2009-12-21 19:58 +0000 [r235944] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) + | 20 lines Merged revisions 235940 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) + | 13 lines Change Monitor to not assume file to write to does not + contain pathing. 227944 changed the fname_base argument to always + append the configured monitor path. This change was necessary to + properly compare files for uniqueness. If a full path is given + though, nothing needs to be appended and that is handled + correctly now. (closes issue #16377) (closes issue #16376) + Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch + uploaded by dant (license 670) ........ ................ + +2009-12-21 17:11 +0000 [r235826] Tilghman Lesher <tlesher@digium.com> + + * /, main/features.c: Merged revisions 235822 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) + | 15 lines Merged revisions 235821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) + | 8 lines Send parking lot announcement to the channel which + parked the call, not the park-ee. (closes issue #16234) Reported + by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded + by tilghman (license 14) 20091221__issue16234__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: yeshuawatso ........ + ................ + +2009-12-20 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz> + + * main/dsp.c: restarts busydetector (if enabled) when DTMF is + received after call is bridged. (closes issue #16389) Reported + by: alecdavis Tested by: alecdavis Patch + dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) + +2009-12-18 23:04 +0000 [r235665] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /, include/asterisk/cdr.h: Merged revisions + 235660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) + | 55 lines Merged revisions 235635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) + | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is + simple in that it reorders the disposition defines so that the + fix for issue 12946 works properly (the default CDR disposition + was changed to AST_CDR_NOANSWER). Also, the + AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all + CDR records are written. The side effects of CDR changes are + scary, so I'm documenting the test cases performed to attempt to + catch any regressions. The following tests were all performed + using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls + B (busy) Hangup C Hangup A (Both SIP and features) A calls B A + blind transfers to C Hangup C (Both SIP and features) A calls B A + attended transfers to C Hangup C A calls B A attended transfers + to C (SIP) C blind transfers to A (features) Hangup A All of the + test scenario CDRs matched. The following tests were performed + just with the patch to ensure proper operation (with + unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten + =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) + (closes issue #16180) Reported by: aatef Patches: bug16180.patch + uploaded by jpeeler (license 325) ........ ................ + +2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 235656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 + (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 + Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion + ........ ................ + + * /, configure, configure.ac: Merged revisions 235573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 + (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 + Dec 2009) | 2 lines Point to the typical missing package, not the + cryptic "termcap support". ........ ................ + +2009-12-17 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 | + file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines + Remove some old code for going to the 'fax' extension when a T.38 + switchover occurs. This would have already happened when we + detected the CNG tone so this was basically a noop. ........ + +2009-12-17 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0 + +2009-12-09 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-rc8 + +2009-12-08 18:33 +0000 [r233731] Tilghman Lesher <tlesher@digium.com> + + * res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) + | 8 lines Find another ref leak and change how we manage module + references. (closes issue #16388) Reported by: parisioa Patches: + 20091208__issue16388.diff.txt uploaded by tilghman (license 14) + Tested by: parisioa, tilghman Review: + https://reviewboard.asterisk.org/r/442/ ........ + +2009-12-08 18:04 +0000 [r233694] Russell Bryant <russell@digium.com> + + * formats/format_sln16.c, formats/format_wav_gsm.c, + formats/format_siren7.c, formats/format_ilbc.c, + formats/format_vox.c, formats/format_pcm.c, + formats/format_h263.c, formats/format_g723.c, + formats/format_h264.c, formats/format_siren14.c, + formats/format_jpeg.c, formats/format_g726.c, + formats/format_gsm.c, formats/format_g729.c, /, + formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) + | 16 lines Set a module load priority for format modules. A + recent change to app_voicemail made it such that the module now + assumes that all format modules are available while processing + voicemail configuration. However, when autoloading modules, it + was possible that app_voicemail was loaded before the format + modules. Since format modules don't depend on anything, set a + module load priority on them to ensure that they get loaded first + when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. + The fix for 1.4 and 1.6.0 will require a different approach since + the module load priority functionality is not present in the + module API. (issue #16412) Reported by: jiddings ........ + +2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com> + + * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6. + +2009-12-07 23:54 +0000 [r233615] Atis Lezdins <atis@iq-labs.net> + + * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 + lines Fix compatibility with valgrind 3.3 and older. (noticed in + issue #16388) Reported by: parisioa Patches: valgrind.supp + uloaded by atis (license 242) Tested by: atis, parisioa ........ + +2009-12-07 23:29 +0000 [r233473-233612] David Vossel <dvossel@digium.com> + + * /, main/utils.c: Merged revisions 233611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | + dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines + fixes incorrect logic in ast_uri_encode issue #16299 ........ + + * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) + | 15 lines Merged revisions 233471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) + | 9 lines fixes missing Contact header angle brackets (closes + issue #16298) Reported by: mgernoth Patches: + reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested + by: dvossel ........ ................ + +2009-12-07 16:16 +0000 [r233396] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | + mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 + lines Do not reject SDP packets describing only non audio + streams. (closes issue #16387) Reported by: zalex1953 Patches: + media-level-c-fix1.diff uploaded by mnicholson (license 96) + Tested by: mnicholson, zalex1953 ........ + +2009-12-04 21:55 +0000 [r233281] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 + (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) + | 7 lines clarify requirecalltoken option in iax.sample.conf + (closes issue #16223) Reported by: bklang Patches: + clarify-iax-requirecalltoken.patch uploaded by bklang (license + 919) ........ ................ + +2009-12-04 21:07 +0000 [r233240] Matthias Nick <mnick@digium.com> + + * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | + mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines + Parse global variables or expressions in hint extensions Parse + global variables or expressions in hint extensions. Like: exten + => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) + Reported by: rmudgett Tested by: mnick, rmudgett ........ + +2009-12-04 17:36 +0000 [r233165] David Vossel <dvossel@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 + (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) + | 6 lines document and rename strip_control() in app_voicemail + (closes issue #16291) Reported by: wdoekes ........ + ................ + +2009-12-04 17:23 +0000 [r233130] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 233100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) + | 14 lines Merged revisions 233092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) + | 7 lines Only do frame payload check for HOLD frames. This code + was added for helping to debug the source of invalid HOLD frames. + However, a side effect of this is that it will incorrectly report + errors for frames that have an integer payload. Make the check + for this block specific to the HOLD frame case. ........ + ................ + +2009-12-04 15:57 +0000 [r233049] Matthias Nick <mnick@digium.com> + + * main/dsp.c, /: Merged revisions 233046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | + 17 lines Merged revisions 233014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | + 11 lines Warning message gets displayed only once Added + additional field 'int display_inband_dtmf_warning', which when + set to '1' displays the warning ('Inband DTMF is not supported on + codec %s. Use RFC2833'), and when set to '0' doesn't display the + warning. Otherwise you would get hundreds of warnings every + second. (closes issue #15769) Reported by: falves11 Patches: + patch_15769_14.txt uploaded by mnick (license 874) Tested by: + mnick, falves11 ........ ................ + +2009-12-03 21:03 +0000 [r232866] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 + (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) + | 8 lines Deprecate "cz" in favor of "cs". Also, change the use + of language codes so that language registers as a prefix, rather + than an exact match. (closes issue #16272) Reported by: patrol-cz + Patches: 20091203__issue16272.diff.txt uploaded by tilghman + (license 14) ........ ................ + +2009-12-03 15:14 +0000 [r232813] David Ruggles <thedavidfactor@gmail.com> + + * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | + diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 + lines Prevent double closing of FDs by EIVR This caused a problem + when asterisk was under heavy load and running both AGI and EIVR + applications. EIVR would close an FD at which point it would be + considered freed and be used by a new AGI instance the second + close would then close the FD now in use by AGI. (closes issue + #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec + Review: https://reviewboard.asterisk.org/r/436/ ........ + +2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com> + + * res/res_musiconhold.c: Oops, really remove it this time + + * res/res_musiconhold.c, /: Recorded merge of revisions + 232660-232661 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 | + tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19 + lines Fix multiple issues with musiconhold, which led to classes + not getting destroyed properly. * Classes are now tracked past + removal from the core container, and module removal is actively + prevented until all references are freed. * A hanging reference + stored in the channel has been removed. This could have caused a + mismatch and the music state not properly cleared, if two or more + reloads occurred between MOH being stopped and MOH being + restarted. * In certain circumstances, duplicate classes were + possible. * A race existed at reload time between a process being + killed and the thread responsible for reading from the related + pipe respawning that process. * Several reference counts have + also been corrected. At least one could have caused deleted + classes to stick around forever, consuming resources. This + originally manifested as MOH external processes that were not + killed at reload time. (closes issue #16279, closes issue #16207) + Reported by: parisioa, dcabot Patches: + 20091202__issue16279__2.diff.txt uploaded by tilghman (license + 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | + 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove + debugging line ........ + +2009-12-02 23:28 +0000 [r232658] David Vossel <dvossel@digium.com> + + * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009) + | 6 lines update CHANGES and UPGRADE.txt for early media behavior + change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: + miki ........ + +2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c, /: Merged revisions 232582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) + | 14 lines Merged revisions 232581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) + | 7 lines Send ack (response/message) after receiving manager + action userevent (closes issue #16264) Reported by: dimas + Patches: event-ack.patch uploaded by dimas (license 88) ........ + ................ + + * main/manager.c, /: Merged revisions 232576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | + jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines + Make manager response to "Action: events" finish with empty line + (closes issue #16275) Reported by: vnovy Patches: manager.c.diff + uploaded by vnovy (license 922) ........ + +2009-12-02 17:11 +0000 [r232359] Joshua Colp <jcolp@digium.com> + + * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | + 12 lines Merged revisions 232355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 + lines Fix a bug where if you hung up very quickly after calling + AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. + (closes issue #16239) Reported by: CGMChris ........ + ................ + +2009-12-02 17:01 +0000 [r232352] David Vossel <dvossel@digium.com> + + * /, main/acl.c: Merged revisions 232351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) + | 12 lines Merged revisions 232350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) + | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in + strace. (closes issue #16290) Reported by: wdoekes ........ + ................ + +2009-12-02 16:43 +0000 [r232348] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | + file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add + support for handling the 415 Unsupported media type response like + we do for a 488 Not acceptable here response. (closes issue + #16186) Reported by: atis Patches: sip_t38_response_415.patch + uploaded by atis (license 242) ........ + +2009-12-02 15:43 +0000 [r232270] David Vossel <dvossel@digium.com> + + * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 + (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) + | 9 lines fixes segfault in func_groupcount closes issue #16337) + Reported by: Parantido Patches: issue_16337.diff uploaded by + dvossel (license 671) Tested by: Parantido, dvossel ........ + ................ + +2009-12-02 14:55 +0000 [r232232] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | + file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix + a bug where a scheduled item ID would get retained on + registrations in a certain scenario causing code to execute + during reload that should not. (issue AST-263) ........ + +2009-12-02 00:52 +0000 [r232094] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 + (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) + | 10 lines Do not modify the gain settings on data calls. (The + digital flag actually represents a data call.) (closes issue + #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt + uploaded by alecdavis (license 585) Tested by: alecdavis ........ + ................ + +2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant <russell@digium.com> + + * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | + russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines + Fix a build error on FreeBSD. ........ + + * /, main/file.c: Merged revisions 232008 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) + | 9 lines Merged revisions 232007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) + | 2 lines Fix a warning pointed out by buildbot. ........ + ................ + +2009-12-01 22:03 +0000 [r231930] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 231927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) + | 19 lines Merged revisions 231911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) + | 12 lines Fix crash with invalid frame data The crash was + happening as a result of a frame containing an invalid data + pointer, but was set with data length of zero. The few times the + issue was reproduced it _seemed_ that the frame was queued + properly, that is the data pointer was set to NULL. I never could + reproduce the crash so as a last resort the crash has been fixed, + but a check in __ast_read has been added to give as much + information about the source of problematic frames in the future. + (closes issue #16058) Reported by: atis ........ ................ + +2009-12-01 21:21 +0000 [r231870] David Vossel <dvossel@digium.com> + + * main/pbx.c, /: Merged revisions 231867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) + | 9 lines Merged revisions 231853 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) + | 3 lines WaitExten m option with no parameters generates frame + with zero datalen but non-null data ptr ........ ................ + +2009-12-01 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-rc7 + +2009-12-01 15:48 +0000 [r231743] Matthew Nicholson <mnicholson@digium.com> + + * /, main/file.c: Merged revisions 231741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec + 2009) | 9 lines Merged revisions 231740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec + 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() + and return an error if no know formats are found. ........ + ................ + +2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming <kpfleming@digium.com> + + * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: + Merged revisions 231692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | + kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 + lines Another round of UDPTL stack fixes/improvements: 1) Allow + users of UDPTL stack to associate a character-string tag with a + UDPTL session, so that log/error/debug messages generated by the + UDPTL stack can be 'connected' to the endpoint that caused them + to be generated. 2) Improve comments (and process) of calculating + the far end's maximum IFP size when redundancy mode is in use for + error correction. 3) When an IFP larger than the calculated 'far + max IFP' size is presented for writing, truncate it rather than + putting in the buffer and allowing the buffer to overflow; this + will cause the ends to retrain to a lower bit rate that produces + IFPs of an appropriate size if possible, and if not possible, the + FAX transfer will fail completely. In these cases, it is due to + the one endpoint supplying a T38FaxMaxDatagram value that is + improperly calculated and is too low to be of use; we have + configuration options available to override this behavior. 4) + Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no + longer needed. ........ + + * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC + 4.4.x happier. + +2009-11-30 21:36 +0000 [r231689] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, + main/app.c: Merged revisions 231688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov + 2009) | 15 lines Merged revisions 231614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov + 2009) | 8 lines Remove duplicate entries from voicemail format + lists. This prevents app_voicemail from entering an infinite loop + when the same format is specified twice in the format list. + (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/429/ ........ + ................ + +2009-11-30 20:47 +0000 [r231605] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | + file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines + When receiving SDP that matches the version of the last one do + not treat it as a fatal error. (closes issue #16238) Reported by: + seandarcy ........ + +2009-11-30 18:57 +0000 [r231505-231558] David Vossel <dvossel@digium.com> + + * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | + dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines + app_queue crashes randomly, often during call-transfers This + patch adds a ref to the queue_ent object's parent call_queue in + queue_exec() so the call_queue won't be destroyed while the the + queue_ent still holds a pointer to it. (closes issue 0015686) + Tested by: dvossel, aragon ........ + + * main/rtp.c, /: Merged revisions 231491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) + | 17 lines Merged revisions 231441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) + | 11 lines fixes crash caused by RTP comfort noise payload + greater than 24 bytes AST-2009-010 (closes issue #16242) Reported + by: amorsen Patches: issue16242.diff uploaded by oej (license + 306) Tested by: amorsen, oej, dvossel ........ ................ + +2009-11-25 22:34 +0000 [r231302] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 231299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) + | 9 lines Merged revisions 231298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) + | 2 lines After a frame duplication failure, unlock the channel + before returning. ........ ................ + +2009-11-25 15:48 +0000 [r231191] Matthew Nicholson <mnicholson@digium.com> + + * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | + mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 + lines Load pbx_lua with global symbols to allow linking with + other lua libraries. Found by Maxim Litnitskiy. ........ + +2009-11-24 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com> + + * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 | + tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines + Found a few places where queue refcounts were counted + incorrectly. Also add debug statements. (closes issue #15982, + closes issue #15984) Reported by: atis Patches: + 20091111__issue15982.diff.txt uploaded by tilghman (license 14) + Tested by: atis ........ + +2009-11-24 18:54 +0000 [r231098] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 231095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | + jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines + Fix erroneous hangup extension execution ast_spawn_extension + behaves differently from 1.4 in that hangups and extensions that + do not exist do not return an error, whereas in 1.6 it does. This + is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN + flag gets set properly. (closes issue #16106) Reported by: + ajohnson Tested by: ajohnson ........ + +2009-11-23 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com> + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 230881 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | + file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines + Change fax detection in chan_sip so it behaves as one would + expect. Internally the way T.38 is negotiated has changed and the + option no longer reflects a behavior that is valid. It will now + look for a CNG tone on received calls and if present send the + call to the 'fax' extension. It is then up to the application or + channel to request the switch over to T.38. ........ + +2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov + 2009) | 9 lines Merged revisions 230839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov + 2009) | 1 line Correct fix for issue #16268... the reporter's + original patch was very close to correct. ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov + 2009) | 12 lines Merged revisions 230772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov + 2009) | 5 lines Ensure that SDP parsing does not ignore the last + line of the SDP. (closes issue #16268) Reported by: sgimeno + ........ ................ + +2009-11-20 22:36 +0000 [r230727] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) + | 7 lines fixes iax2 show cache locking error, thanks alecdavis! + (closes issue #16094) Reported by: alecdavis Patches: + bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: + alecdavis, dvossel ........ + +2009-11-20 21:07 +0000 [r230629] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 230628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov + 2009) | 15 lines Merged revisions 230627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov + 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR + if it exists. This is necessary for the recordagentcalls option + in chan_agent to store the recorded file name in the bridge CDR. + (closes issue #14590) Reported by: msetim Patches: + queue_agent_userfield.patch uploaded by Laureano (license 265) + Tested by: Laureano, mnicholson ........ ................ + +2009-11-20 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com> + + * main/audiohook.c, /, include/asterisk/audiohook.h: Merged + revisions 230583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | + dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines + audiohook signal trigger on every status change (issue #14618) + Review: https://reviewboard.asterisk.org/r/434/ ........ + + * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 + (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) + | 10 lines fixes MixMonitor thread not exiting when + StopMixMonitor is used (closes issue #16152) Reported by: AlexMS + Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license + 671) Tested by: dvossel, AlexMS Review: + https://reviewboard.asterisk.org/r/424/ ........ ................ + +2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 | + kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line + Fix another buglet in T.38 session teardown at the end of FAX + sessions. ........ + + * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 | + kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 + lines Ensure that only one end of a T.38 session initiates + teardown at completion. ........ + + * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600 + (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov + 2009) | 6 lines Correct mistaken option name in error message. + The configuration option for allowing hosts to make + non-token-based calls is 'calltokenoptional', not + 'calltokenignore'. (reported on asterisk-users) ........ + ................ + +2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | + 15 lines Merged revisions 230144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 + lines Respect the maddr parameter in the Via header. (closes + issue #14446) Reported by: frawd Patches: via_maddr.patch + uploaded by frawd (license 610) Tested by: frawd ........ + ................ + + * channels/chan_local.c, /: Merged revisions 230039 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, + 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 + lines Fix a crash caused by two threads thinking they should both + free the chan_local private structure when only one should. + (closes issue #15314) Reported by: sroberts Patches: + Issue15314_Move_Nulling_owner.patch uploaded by davidw (license + 780) Tested by: davidw, lottc ........ ................ + + * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged + revisions 229966 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | + 13 lines Merged revisions 229965 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 + lines Document a limitation in the AVAILSTATUS variable from + ChanIsAvail and provide a workaround for it that does not change + existing behavior. (closes issue #14426) Reported by: macli + ........ ................ + +2009-11-13 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-rc6 + +2009-11-13 15:57 +0000 [r229915] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | + file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix + T.38 negotiation regression introduced with the SDP parser + changes. ........ + +2009-11-12 23:31 +0000 [r229752] Jason Parker <jparker@digium.com> + + * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 | + qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix + mute toggling on OSS channels. ........ + +2009-11-12 16:47 +0000 [r229671] David Vossel <dvossel@digium.com> + + * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 + (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) + | 6 lines fixes merging error, datastore was being freed in the + wrong function. (closes issue #16219) Reported by: aragon + ........ ................ + +2009-11-11 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com> + + * doc/externalivr.txt: Merged revisions 229568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 | + diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9 + lines Remove non-functional feature from ExternalIVR + documentation Remove non-functional socket implementation of + ExternalIVR from documentation (closes issue #16225) Reported by: + thedavidfactor Patches: externalivr.txt.20091111.1542.patch + uploaded by thedavidfactor (license 903) ........ + +2009-11-11 19:56 +0000 [r229492-229502] David Brooks <dbrooks@digium.com> + + * main/pbx.c, /: Merged revisions 229499 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) + | 15 lines Merged revisions 229498 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) + | 8 lines Solaris doesn't like NULL going to ast_log Solaris will + crash if NULL is passed to ast_log. This simple patch simply uses + S_OR to get around this. (closes issue #15392) Reported by: + yrashk ........ ................ + + * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) + | 7 lines Flags not initialized in app_softhangup.c, causing + undefined behavior Trivial patch [kobaz] to initialize an + ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ + +2009-11-10 22:17 +0000 [r229366] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 229361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) + | 19 lines Merged revisions 229360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) + | 12 lines If two pattern classes start with the same digit and + have the same number of characters, they will compare equal. The + example given in the issue report is that of [234] and [246], + which have these characteristics, yet they are clearly not + equivalent. The code still uses these two characteristics, yet + when the two scores compare equal, an additional check will be + done to compare all characters within the class to verify + equality. (closes issue #15421) Reported by: jsmith Patches: + 20091109__issue15421__2.diff.txt uploaded by tilghman (license + 14) Tested by: jsmith, thedavidfactor ........ ................ + +2009-11-10 22:04 +0000 [r229359] David Ruggles <thedavidfactor@gmail.com> + + * doc/externalivr.txt: Merged revisions 229356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov + 2009) | 16 lines Merged revisions 229355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov + 2009) | 9 lines Fix ExternalIVR Documentation Remove + documentation for event that doesn't function (closes issue + #16220) Reported by: thedavidfactor Patches: + externalivr.txt.20091110.1622.patch uploaded by thedavidfactor + (license 903) ........ ................ + +2009-11-10 21:33 +0000 [r229354] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | + tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines + When GOSUB is invoked within an AGI, it may not exit correctly. + (closes issue #16216) Reported by: atis Patches: + 20091110__atis_work.diff.txt uploaded by tilghman (license 14) + Tested by: atis ........ + +2009-11-10 20:09 +0000 [r229285] Joshua Colp <jcolp@digium.com> + + * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | + 15 lines Merged revisions 229281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 + lines Remove broken support for direct transcoding between G.726 + RFC3551 and G.726 AAL2. On some systems the translation core + would actually consider g726aal2 -> g726 -> signed linear to be a + quicker path then g726aal2 -> signed linear which exposed this + problem. (closes issue #15504) Reported by: globalnetinc ........ + ................ + +2009-11-10 17:52 +0000 [r229232] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 + (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) + | 9 lines don't crash on log message in solaris AST-2009-006 + (closes issue #16206) Reported by: bklang Tested by: bklang + ........ ................ + +2009-11-10 17:39 +0000 [r229231] David Ruggles <thedavidfactor@gmail.com> + + * doc/externalivr.txt: Merged revisions 229228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov + 2009) | 18 lines Merged revisions 229191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov + 2009) | 11 lines Document ExternalIVR event tag collision + ExternalIVR uses the D tag for two different event types. This + documents that behavior and how to differentiate between the two + cases. Also includes a minor spelling fix and clarification + (closes issue #16211) Reported by: thedavidfactor Patches: + externalivr.txt.20091109.1507.patch uploaded by thedavidfactor + (license 903) ........ ................ + +2009-11-10 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com> + + * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset + props that were accidently deleted in 229088. + +2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com> + + * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009) + | 11 lines fixes pgsql double free of threadstorage A thread + storage variable was being freed incorrectly, which resulted in a + double free if two queries were made in the same thread. (closes + issue #16011) Reported by: cristiandimache Patches: + issue16011.diff uploaded by dvossel (license 671) ........ + +2009-11-10 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com> + + * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c, + UPGRADE.txt: Reverted revision 202007. (closes issue #16175) + Reported by: paul-tg Tested by: paul-tg + +2009-11-10 11:25 +0000 [r229078] Gavin Henry <ghenry@suretecsystems.com> + + * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 + Nov 2009) | 20 lines Schema file additions * Added + AsteriskDialplan, AsteriskAccount and AsteriskMailbox + objectClasses to allow standalone dialplan, account and mailbox + entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, + AstAccountTransport, AstAccountPromiscRedir, - + AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, + - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed + redundant IPaddr (there's already IPAddress) - Gives more + configuration Flags for SIP-Users available (tested) - Allows to + create Asterisk Attributes in defined Asterisk ObjectClasses + without extensibleObject (which really should be the last + resort); gives also additional possibilities for LDAP-filter + (closes issue #15874) Reported by: Medozas Patches: + asterisk.ldap-schema.patch uploaded by Medozas (license 41) + Tested by: Medozas, suretec ........ + +2009-11-09 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com> + + * channels/chan_local.c, /: Merged revisions 229015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) + | 8 lines Don't crash when bridge->tech_pvt == NULL This is a + similar solution to what is in place for chan_agent (closes issue + #16003) Reported by: atis Tested by: twilson ........ + +2009-11-09 22:17 +0000 [r229012] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes segfault when transferring a queue + caller In sip_hangup we attempted to lock p->owner after we set + it to NULL. Thanks to fhackenberger for reporting the issue and + submitting a patch. (closes issue #15848) Reported by: + fhackenberger Patches: digium_bug_0015848 uploaded by + fhackenberger (license 592) Tested by: fhackenberger, lmadsen, + TomS, shin-shoryuken, dvossel + +2009-11-09 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-rc5 + +2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com> + + * main/channel.c: Merged revisions 228897 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) + | 14 lines Merged revisions 228896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) + | 6 lines Update WARNING message. Update a WARNING message to + give a suggested fix when encountered. (closes issue #16198) + Reported by: atis Tested by: atis ........ ................ + +2009-11-09 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 + (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov + 2009) | 8 lines Perform limited bounds checking when destroying + ast_mutex_t structures to make sure we don't try to use negative + indices. (closes issue #15588) Reported by: zerohalo Patches: + 20090820__issue15588.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ + +2009-11-06 22:37 +0000 [r228694] David Vossel <dvossel@digium.com> + + * main/channel.c, /: Merged revisions 228693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) + | 16 lines Merged revisions 228692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) + | 9 lines fixes audiohook write crash occuring in chan_spy + whisper mode. After writing to the audiohook list in ast_write(), + frames were being freed incorrectly. Under certain conditions + this resulted in a double free crash. (closes issue #16133) + Reported by: wetwired ........ ................ + +2009-11-06 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com> + + * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600 + (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov + 2009) | 8 lines Properly handle '=' while decoding base64 + messages and null terminate strings returned from BASE64_DECODE. + (closes issue #15271) Reported by: chappell Patches: + base64_fix.patch uploaded by chappell (license 8) Tested by: + kobaz ........ ................ + +2009-11-06 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | + 11 lines Merged revisions 228547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 + lines Don't overwrite caller ID name on a trunk with the + configured fullname when using users.conf (issue ABE-1989) + ........ ................ + +2009-11-06 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.2.0-rc4 + +2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com> + + * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 + lines Fix the localchannel.tex file. ........ + +2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com> + + * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 | + dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines + Fixes merging issue from 1.4, frame data is held in data.ptr in + trunk ........ + + * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) + | 19 lines Merged revisions 228418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) + | 13 lines fixes segfault in iLBC For reasons not yet known, it + appears possible for an ast_frame to have a datalen greater than + zero while the actual data is NULL during Packet Loss + Concealment. Most codecs don't support PLC so this doesn't affect + them. This patch catches the malformed frame and prevents the + crash from occuring. Additional efforts to determine why it is + possible for a frame to look like this are still being + investigated. (issue #16979) ........ ................ + +2009-11-06 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com> + + * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | + 14 lines Merged revisions 228409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 + lines Fix a bug caused by a partially invalid frame (from the + jitterbuffer) passing through the Asterisk core. (closes issue + #15560) Reported by: jvandal (closes issue #15709) Reported by: + covici ........ ................ + +2009-11-06 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com> + + * /, main/astfd.c: Merged revisions 228339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) + | 12 lines Merged revisions 228338 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) + | 5 lines fixes crash in astfd.c (closes issue #15981) Reported + by: slavon ........ ................ + + * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 + Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c + (closes issue #15394) Reported by: boroda Patches: + bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) + Tested by: dbrooks, boroda ........ + +2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 | + tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines + Yet another error message in the dialplan (thanks, + rmudgett/russellb) ........ + +2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | + jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines + Fix the fix for chanspy option o In 224178, I assumed the + uploaded patch was correct as it had received positive feedback. + The flags were being checked in the incorrect location. Upon + testing the fix this time it was also found that the flags from + the dialplan weren't being copied to the + chanspy_translation_helper. (closes issue #16167) Reported by: + marhbere ........ + +2009-11-05 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 | + tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines + MEETME_INFO should not return a literal error message to the + dialplan. (closes issue #15450) Reported by: JimVanM Patches: + meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested + by: JimVanM ........ + +2009-11-05 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 + (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) + | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to + chan_misdn connection. Patch submitted by gknispel_proformatique, + tested by francesco_r. "I have many crash since i have upgraded + to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out + an ast_frame. (closes issue #16041) Reported by: francesco_r + ........ ................ + +2009-11-05 19:20 +0000 [r228093] Jason Parker <jparker@digium.com> + + * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 + (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | + 8 lines Fix crash on VPB exception when no hardware is present. + (closes issue #14970) Reported by: tzafrir Patches: + vpb_exception.diff uploaded by tzafrir (license 46) Tested by: + markwaters ........ ................ + +2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com> + + * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) + | 4 lines Don't crash if no arguments are passed. (closes issue + #16119) Reported by: thedavidfactor ........ + +2009-11-04 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) + | 21 lines Merged revisions 227944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) + | 14 lines Fix incorrect filename comparsion after monitor file + change The logic to detect if a requested file is indeed a + different file from the current file was incorrect. The main + issue being confusion of the use of filename_base which was + previously set without pathing information and then compared to + another full path. Robust file comparison logic has been added to + properly check if two files are the same even if symlinks are + used. (closes issue #15313) Reported by: caspy Patches: + 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license + 325) but mostly tilghman's work ........ ................ + +2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov + 2009) | 17 lines Merged revisions 227827 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov + 2009) | 10 lines This patch modifies the Dial application to + monitor the calling channel for hangups while playing back + announcements. (closes issue #16005) Reported by: falves11 + Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson + (license 96) Tested by: mnicholson, falves11 Review: + https://reviewboard.asterisk.org/r/407/ ........ ................ + + * channels/chan_sip.c: Modify the SDP parsing code to parse session + and media level items separately. With the new code, media level + proprieties should no longer be confused with session level + proprieties. This change also reorganizes some of the SDP parsing + code which should make it easier to manage in the future. (closes + issue #14994) Reported by: frawd + +2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com> + + * /, static-http/prototype.js: Merged revisions 227739 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, + 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 + lines Fix a security issue where it may be possible for someone + to execute a cross-site AJAX request exploit. (AST-2009-009) + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | + 12 lines Merged revisions 227700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 + lines Fix a security issue where sending a REGISTER with a + differing username in the From URI and Authorization header would + reveal whether it was valid or not. (AST-2009-008) ........ + ................ + +2009-11-03 20:01 +0000 [r227375] Jason Parker <jparker@digium.com> + + * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | + 9 lines Fix some build issues on Solaris. (closes issue #14517) + (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded + by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell + ........ + +2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen <lmadsen@digium.com> + + * apps/app_controlplayback.c, /: Merged revisions 227368 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 + Nov 2009) | 8 lines Change warning message to debug message. + app_controlplayback outputs a warning, when in fact it is normal. + (closes issue #16071) Reported by: atis Patches: + controlplayback_warning.patch uploaded by atis (license 242) + ........ + + * configs/extensions.conf.sample, /: Merged revisions 227361 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 + Nov 2009) | 11 lines Additional fixes to the + extensions.conf.sample file. Update the extensions.conf.sample + [stdexten] context so that we use the variable instead of + requiring it to be passed explicitly. Also updated uses of the + [stdexten] context throughout. (closes issue #15858) Reported by: + pprindeville Patches: stdexten-context-update.txt uploaded by + lmadsen (license 10) Tested by: pprindeville ........ + +2009-11-03 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) + | 4 lines Make sure the outgoing flag is cleared if a new channel + fails to get created for outgoing calls. This is the relevant + portion of asterisk/trunk -r226648 ........ + +2009-11-03 17:14 +0000 [r227239] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 | + dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines + user.conf entries in SIP were not having their peer type set. + (closes issue #16120) Reported by: jsmith ........ + +2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | + 12 lines Merged revisions 227166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 + lines Fix a bug where an RPID header could be generated with a + blank username in the URI. (closes issue #15909) Reported by: + kobaz ........ ................ + +2009-11-03 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 227162 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 + Nov 2009) | 7 lines Update extensions.conf.sample file to fix + incorrect extensions. (closes issue #15857) Reported by: + pprindeville Patches: stdexten.patch#2 uploaded by pprindeville + (license 347) Tested by: pprindeville ........ + +2009-11-03 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net> + + * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, + 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 + lines Use proper response code when violating Contact ACL's. + https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a + quick review. (EDVX-003) ........ ................ + +2009-11-02 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com> + + * channels/chan_sip.c: SIP channel name uniqueness SIP channel + names were supposed to be unique by way of a name suffix derived + from the pointer to the channel's private data. Uniqueness was + preserved on 32-bit systems, but not on 64-bit systems. This + patch, as suggested by kpfleming, replaces this suffix with a + simple incremented unsigned int. (closes issue #15152) Reported + by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ + +2009-11-02 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | + 18 lines Merged revisions 226889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | + 11 lines Fix a bug where the recorded privacy introduction file + would not get removed if the caller hung up while the called + party had not yet answered. This was fixed by introducing an + argument to the 'n' option which, when enabled, removes the + introduction file under all scenarios. This was done to preserve + the behavior that has existed for quite some time. (closes issue + #14674) Reported by: ulogic Patches: bug14674.patch uploaded by + jpeeler (license 325) ........ ................ + +2009-11-02 17:17 +0000 [r226815] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 + (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) + | 8 lines Don't allow two separate instances of safe_asterisk + when restarting from the init script. (closes issue #14562) + Reported by: davidw Patches: Initially + 20091022__issue14562.diff.txt uploaded by tilghman (license 14) + Modified to 20091030__Issue14562_diff.txt uploaded by davidw + (license 780) Tested by: davidw ........ ................ + +2009-10-29 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com> + + * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged + revisions 226532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | + 13 lines Merged revisions 226531 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 + lines Add an option to enabling passing music on hold start and + stop requests through instead of acting on them in chan_local. + (closes issue #14709) Reported by: dimas ........ + ................ + +2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * build_tools/get_documentation, /: remove empty awk pattern (//) + Solaris 10 nawk doesn't like the empty pattern such as '//' for + 'always'. Just remove that. No pattern at all always matches. + Merged revisions 226453 via svnmerge from + http://svn.digium.com/svn/asterisk/trunk + +2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com> + + * configs/sip.conf.sample: Merged revisions 226384 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 + (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) + | 9 lines Update documentation in sip.conf.sample. Update the + documentation in sip.conf.sample in order to make it more clear + that directmedia/canreinvite do not cause Asterisk to ignore + reINVITEs. It is only used to stop Asterisk from generating a + reINVITE, but does not stop it from accepting them if necessary. + (closes issue #15644) Reported by: lmadsen ........ + ................ + + * doc/tex/channelvariables.tex: Merged revisions 226378 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 + (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) + | 7 lines Update CALLINGSUBADDR channel variable documentation. + (closes issue #15734) Reported by: alecdavis Patches: + channelvariables.tex.diff.txt uploaded by alecdavis (license 585) + Tested by: alecdavis ........ ................ + +2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/linkedlists.h: Merged revisions 226305 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 + (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 + Oct 2009) | 2 lines Fix documentation (pointed out by + TheDavidFactor on #-dev) ........ ................ + + * main/manager.c, /: Merged revisions 226159 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) + | 14 lines Merged revisions 226138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) + | 7 lines Manager output is not always NULL-terminated, so force + a NULL at the end of the filestream. (closes issue #15495) + Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded + by tilghman (license 14) Tested by: pdf ........ ................ + +2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com> + + * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 | + twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines + Don't prepend the URI prefix to the post directory ........ + +2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * /, configure, configure.ac: detect ARM Linux EABI OSARCH as + linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even + if host_os is linux-gnueabi * When checking if we are Linux, + check OSARCH rather than host_os The newer ARM ABI ("EABI") shows + the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch + sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is + tested for the value of 'linux-gnu' in one or two places in the + tree. This patch also fixes the check libcap to check for $OSARCH + rather than $host_os . See also: + http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via + svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 + Merged revisions 226018 via svnmerge from + http://svn.digium.com/svn/asterisk/trunk + +2009-10-26 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 | + jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines + ACL check not present for verifying SIP INVITEs The ACL check in + check_peer_ok was missing and has now been restored. The missing + check allowed for calls to be made on prohibited networks where + an ACL was defined in sip.conf and the allowguest option was set + to off. See the AST security advisory below for more information. + Merge code associated with AST-2009-007. (closes issue #16091) + Reported by: thom4fun ........ + +2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_fax.c: Backport audio handling loop fixes from trunk + version of app_fax. This backport resolves some issues handling + audio frames during FAX processing, and ensures that the FAX + application doesn't accidentally get notified of a T.38 + switchover at the end of a successful FAX. (closes issue #16127) + +2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 | + dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines + Fixes an iterator memory leak and uninitialized memory ........ + +2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 225582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct + 2009) | 17 lines Merged revisions 225581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct + 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on + every build. For some reason the menuselect.makeopts file was + listed as PHONY in the Makefile, resulting in 'make' needing to + rebuild it for every build. This then resulted in the embedded + module rules being rebuilt on every build, which can be slow and + is unnecessary. This patch fixes the problem by properly allowing + 'make' to know when the menuselect.makeopts file needs to be + rebuilt (defining the proper dependencies). ........ + ................ + +2009-10-22 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com> + + * README, /: Merged revisions 225515 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 | + lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines + Update README documentation. Update the README documentation to + correctly describe which CLI command you should use when + attempting to get help from the CLI. (closes issue #16064) + Reported by: thedavidfactor Patches: readme.patch uploaded by + thedavidfactor (license 903) ........ + +2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com> + + * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c, + /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | + dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines + SIP TCP/TLS: move client connection setup/write into tcp helper + thread, various related locking/memory fixes. What this patch + fixes 1.Moves sip TCP/TLS connection setup into the TCP helper + thread: Connection setup takes awhile and before this it was + being done while holding the monitor lock. 2.Moves TCP/TLS + writing to the TCP helper thread: Through the use of a packet + queue and an alert pipe, the TCP helper thread can now be woken + up to write data as well as read data. 3.Locking error: sip_xmit + returned an XMIT_ERROR without giving up the tcptls_session lock. + This lock has been completely removed from sip_xmit and placed in + the new sip_tcptls_write() function. 4.Memory leak: When creating + a tcptls_client the tls_cfg was alloced but never freed unless + the tcptls_session failed to start. Now the session_args for a + sip client are an ao2 object which frees the tls_cfg on + destruction. 5.Pointer to stack variable: During + sip_prepare_socket the creation of a client's + ast_tcptls_session_args was done on the stack and stored as a + pointer in the newly created tcptls_session. Depending on the + events that followed, there was a slight possibility that pointer + could have been accessed after the stack returned. Given the new + changes, it is always accessed after the stack returns which is + why I found it. Notable code changes 1.I broke tcptls.c's + ast_tcptls_client_start() function into two functions. One for + creating and allocating the new tcptls_session, and a separate + one for starting and handling the new connection. This allowed me + to create the tcptls_session, launch the helper thread, and then + establish the connection within the helper thread. 2.Writes to a + tcptls_session are now done within the helper thread. This is + done by using an alert pipe to wake up the thread if new data + needs to be sent. The thread's sip_threadinfo object contains the + alert pipe as well as the packet queue. 3.Since the threadinfo + object contains the alert pipe, it must now be accessed outside + of the helper thread for every write (queuing of a packet). For + easy lookup, I moved the threadinfo objects from a linked list to + an ao2_container. (closes issue #13136) Reported by: pabelanger + Tested by: dvossel, whys (closes issue #15894) Reported by: + dvossel Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/380/ ........ + +2009-10-22 21:54 +0000 [r225488] Leif Madsen <lmadsen@digium.com> + + * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions + 225485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) + | 19 lines Merged revisions 225484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) + | 11 lines Clean valgrind output by suppressing false errors. + Update valgrind.txt documentation and add valgrind.supp file in + order to allow those who are creating valgrind output to have + less false errors in the logfile. (closes issue #16007) Reported + by: atis Patches: valgrind.txt.diff uploaded by atis (license + 242) asterisk2.supp uploaded by atis (license 242) Tested by: + atis, amorsen ........ ................ + +2009-10-22 17:14 +0000 [r225363] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: + Merged revisions 225360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) + | 11 lines Merged revisions 225105 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) + | 4 lines Fix documentation for ast_softhangup() and correct the + misuse thereof. (closes issue #16103) Reported by: majorbloodnok + ........ ................ + +2009-10-21 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 + (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) + | 13 lines IAX2: VNAK loop caused by signaling frames with no + destination call number It is possible for the PBX thread to + queue up signaling frames before a destination call number is + received. This can result in signaling frames being sent out with + no destination call number. Since recent versions of Asterisk + require accurate destination callnumbers for all Full Frames, + this can cause a VNAK loop to occur. To resolve this no signaling + frames are sent until a destination callnumber is received, and + destination call numbers are now only required for iax_pvt + matching when the frame is an ACK. Review: + https://reviewboard.asterisk.org/r/413/ ........ ................ + + * configs/sip.conf.sample, channels/chan_iax2.c, + configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions + 225033 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) + | 27 lines Merged revisions 225032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) + | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller + id removes '(', ' ', ')', non-trailing '.', and '-' from the + string. This means values such as 555.5555 and test-test result + in 555555 and testtest. There are instances, such as Skype + integration, where a specific value is passed via caller id that + must be preserved unmodified. This patch makes the shrinking of + caller id optional in chan_sip and chan_iax in order to support + such cases. By default this option is on to preserve previous + expected behavior. (closes issue #15940) Reported by: dimas + Patches: v2-15940.patch uploaded by dimas (license 88) + 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) + Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/408/ ........ ................ + +2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com> + + * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 + (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) + | 5 lines Pay attention to the return value of the manipulate + function. While this looks like an optimization, it prevents a + crash from occurring when used with certain audiohook callbacks + (diagnosed with SVN trunk, backported to 1.4 to keep the source + consistent across versions). ........ ................ + +2009-10-20 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 224774 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | + 12 lines Merged revisions 224773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 + lines Add support for relaying early media in the features + attended transfer option. (closes issue #14828) Reported by: + licedey ........ ................ + +2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com> + + * main/rtp.c, /: Merged revisions 224671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct + 2009) | 14 lines Merged revisions 224670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct + 2009) | 7 lines Correct timestamp calculations when RTP sample + rates over 8kHz are used. While testing some endpoints that + support 16kHz and 32kHz sample rates, some log messages were + generated due to calc_rxstamp() computing timestamps in a way + that produced odd results, so this patch sanitizes the result of + the computations. ........ ................ + +2009-10-19 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | + 12 lines Merged revisions 224565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 + lines Do not attempt early media bridging (ie: direct RTP setup) + if options are enabled that should prevent it. (closes issue + #14763) Reported by: cupotka ........ ................ + +2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com> + + * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct + 2009) | 1 line Remove useless debugging message. ........ + +2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) + | 3 lines Allow ODBC storage to be queried with multiple + mailboxes, and remove multiple goto's. This corrects an issue + reported on the -users list. ........ + + * configs/res_odbc.conf.sample, /: Merged revisions 224446 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18 + Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias + for "autocommit", but instead controls the default disposition of + uncommitted transactions. ........ + +2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 + (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) + | 13 lines Fix stale caller id data from being reported in AMI + NewChannel event The problem here is that chan_dahdi is designed + in such a way to set certain values in the dahdi_pvt only once. + One of those such values is the configured caller id data in + chan_dahdi.conf. For PRI, the configured caller id data could be + overwritten during a call. Instead of saving the data and + restoring, it was decided that for all non-analog channels it was + simply best to not set the configured caller id in the first + place and also clear it at the end of the call. (closes issue + #15883) Reported by: jsmith ........ ................ + +2009-10-16 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 + (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) + | 18 lines Never released PRI channels when using Busy() or + Congestion() dialplan apps. When the Busy() or Congestion() + application is used towards ISDN (an ISDN progress is sent), the + responding ISDN Disconnect or Release may contain the ISDN cause + user busy or one of the congestion causes. In chan_dahdi.c these + causes will only set the needbusy or needcongestion flags and not + activate the softhangup procedure. Unfortunately only the latter + can interrupt the endless wait loop of Busy()/Congestion(). + Result: PRI channels staying in state busy for the rest of + asterisk life or until the other end times out and forces the + call to clear. (in issue 0014292) Reported by: tomaso Patches: + disc_rel_userbusy.patch uploaded by tomaso (license 564) (This + patch is unrelated to the issue.) ........ ................ + +2009-10-15 15:58 +0000 [r224181] Jeff Peeler <jpeeler@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | + jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines + Readd removed ability to allow listening to one side of the call + in app_chanspy (Option o) (closes issue #15675) Reported by: + john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks + (license 790) Tested by: jgutierrez on users list: + http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html + ........ + +2009-10-12 23:55 +0000 [r223835] Jeff Peeler <jpeeler@digium.com> + + * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) + | 15 lines Merged revisions 223804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) + | 8 lines Ensure ringing continues for branched calls after + progress is received While waiting for an answer, don't send + progress for branched calls for which ringing was sent. (closes + issue #15028) Reported by: fnordian ........ ................ + +2009-10-12 21:01 +0000 [r223757] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) + | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 + options SWP-151 ........ + +2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 + Oct 2009) | 13 lines Remove automatic switching from T.38 to + voice mode in chan_sip. chan_sip has some code to automatically + switch from T.38 mode to voice mode when a voice frame is written + to the channel while it is in T.38 mode; this was intended to + handle the situation when a FAX transmission has ended and the + channel is not yet hung up, but is causing problems at the + beginning of FAX sessions as well when there are still voice + frames 'in flight' at the time the T.38 negotiation completes. + This patch removes the automatic switchover, and changes app_fax + to explicitly switch off T.38 mode when the FAX transmission + process ends. (closes issue #16025) Reported by: jamicque + ........ + +2009-10-11 17:32 +0000 [r223490] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 223487 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) + | 17 lines Merged revisions 223485-223486 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) + | 6 lines Don't use data outside of its scope. The purpose of + this code was to have a hangup frame put on the list of deferred + frames. However, the code that read the hangup frame was outside + of the scope of where the hangup frame was declared. ........ + r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) + | 2 lines Remove some unnecessary code. ........ ................ + +2009-10-09 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation + of PRIREDIRECTIONREASON set by chan_sip. This commit is the + simplest way to solve a problem that has already been solved in + trunk with the "COLP/CONP and Redirecting party information into + Asterisk" commit. In trunk the redirection reason is translated + into a generic redirect reason. I would have had to do the same + fix except chan_sip never reads PRIREDIRECTREASON. So both + chan_dahdi and chan_h323 have been modified to interpret the one + different redirect reason of "no-answer" properly and set the + ISDN reason code 2 of "no reply". (closes issue #15033) Reported + by: steinwej + +2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | + kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 + lines Initiate T.38 switchover when acting as called party, + regardless of FAX direction. SendFAX() and ReceiveFAX() can be + given options to indicate whether they should act as the calling + or called party; this mode should be used to decide whether to + initiate a switchover to T.38, not the direction that the FAX + transfer will take place. (closes issue #16039) Reported by: + jamicque ........ + +2009-10-09 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com> + + * main/channel.c, /: Merged revisions 223273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct + 2009) | 14 lines Merged revisions 223225 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct + 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING + when originating calls. (closes issue #15104) Reported by: + nblasgen Patches: manager-timeout1.diff uploaded by mnicholson + (license 96) Tested by: nblasgen, mnicholson ........ + ................ + +2009-10-09 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct + 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, + 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c + ........ ................ + +2009-10-09 17:55 +0000 [r223208] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) + | 16 lines Merged revisions 223205 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) + | 10 lines fixes sip registration using authuser in user.conf + (closes issue #14954) Reported by: tornblad Tested by: + mmichelson, tornblad, dvossel ........ ................ + +2009-10-09 17:27 +0000 [r223173] Matthew Nicholson <mnicholson@digium.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct + 2009) | 8 lines Don't close the sqlite database when reloading. + Only close the database when unloading. (closes issue #15953) + Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by + frawd (license 610) Tested by: frawd ........ + +2009-10-09 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | + dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines + 'auth=' did not parse md5 secret correctly (closes issue #15949) + Reported by: ebroad Patches: authparsefix.patch uploaded by + ebroad (license 878) 15949_trunk.diff uploaded by dvossel + (license 671) Tested by: ebroad ........ + + * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | + dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines + p->peerauth is always empty in transmit_register() When using + callbackextension or specifing the peer name in a registration + string, the peer's specific auth settings set by the "auth=" + strings within the peer definition are not used by the + registration. Thanks to ebroad for reporting the issue and + providing the patch. (closes issue #15955) Reported by: ebroad + Patches: regauthfix.patch uploaded by ebroad (license 878) + ........ + +2009-10-08 20:00 +0000 [r222883] Russell Bryant <russell@digium.com> + + * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, + /, main/file.c: Merged revisions 222880 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) + | 51 lines Merged revisions 222878 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) + | 44 lines Make filestream frame handling safer by isolating + frames before returning them. This patch is related to a number + of issues on the bug tracker that show crashes related to freeing + frames that came from a filestream. A number of fixes have been + made over time while trying to figure out these problems, but + there re still people seeing the crash. (Note that some of these + bug reports include information about other problems. I am + specifically addressing the filestream frame crash here.) I'm + still not clear on what the exact problem is. However, what is + _very_ clear is that we have seen quite a few problems over time + related to unexpected behavior when we try to use embedded frames + as an optimization. In some cases, this optimization doesn't + really provide much due to improvements made in other areas. In + this case, the patch modifies filestream handling such that the + embedded frame will not be returned. ast_frisolate() is used to + ensure that we end up with a completely mallocd frame. In + reality, though, we will not actually have to malloc every time. + For filestreams, the frame will almost always be allocated and + freed in the same thread. That means that the thread local frame + cache will be used. So, going this route doesn't hurt. With this + patch in place, some people have reported success in not seeing + the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) + Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt + uploaded by russell (license 2) Tested by: aragon, russell + (closes issue #15817) Reported by: zerohalo Tested by: zerohalo + (closes issue #15845) Reported by: marhbere Review: + https://reviewboard.asterisk.org/r/386/ ........ ................ + +2009-10-08 19:41 +0000 [r222874] David Vossel <dvossel@digium.com> + + * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions + 222873 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | + dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines + fixes an ast_netsock_list memory leak. ABE-1998 Review: + https://reviewboard.asterisk.org/r/395/ ........ + +2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 + (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) + | 12 lines Fix memory leak if chan_misdn config parameter is + repeated. Memory leak when the same config option is set more + than once in an misdn.conf section. Why must this be considered? + Templates! Defining a template with default port options and + later adding to or overriding some of them. Patches: + memleak-misdn.patch JIRA ABE-1998 ........ ................ + + * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 + (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) + | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: + astdtmf must be set to "yes". With "no", buffer loss does not + occur. The translated frame "f2" when passing through + ast_dsp_process() is not freed whenever it is not used further in + process_ast_dsp(). Then in the end it is never ever freed. + Patches: translate.patch JIRA ABE-1993 ........ ................ + +2009-10-07 18:06 +0000 [r222549] Jason Parker <jparker@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 222548 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct + 2009) | 5 lines Remove 'keepstats' queue option from sample + config, as it's no longer used. + https://reviewboard.asterisk.org/r/115/ (closes issue #15820) + Reported by: kshumard ........ + +2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com> + + * funcs/func_strings.c: Fix merge error. + +2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) + | 14 lines Merged revisions 222542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) + | 8 lines crash on transfer handle_invite_replaces() attempts to + uplock a pvt's owner channel without first verifing that it + exists. (issue #16027) ........ ................ + +2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 + (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) + | 8 lines Add missing unlock(s) in dahdi_read (two cases in + trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis + ........ ................ + + * channels/chan_dahdi.c: Fix potential crash when entire span + request is received. The variable index used in this scenario for + accessing the dahdi_pvts was wrong and was most likely copied + from the several other places it is used correctly. (closes issue + #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch + uploaded by tsearle (license 373) + + * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) + | 9 lines Fix 222298 (crash during destruction of second channel + when variable set with setvar). I mistakenly reasoned that setvar + would be used on all channels. Since it can be set per channel, + give each dahdi channel a copy of the variable. (related to + #15899) ........ + +2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions + 222309 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 | + tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10 + lines Change schema query to involve the use of an optional + schema parameter. This change is done in such a way as to allow + the driver to continue to function with older databases which + don't have these features. (closes issue #16000) Reported by: + jamicque Patches: 20091002__issue16000.diff.txt uploaded by + tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + +2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) + | 9 lines Fix crash during destruction of second channel when + variable set with setvar. The setvar line in chan_dahdi.conf is + shared among all the channels, so make sure to only free the + resources only when the last channel is destroyed. (closes issue + #15899) Reported by: tzafrir ........ + +2009-10-06 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com> + + * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | + tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines + When we call a gosub routine, the variables should be scoped to + avoid contaminating the caller. This affected the ~~EXTEN~~ hack, + where a subroutine might have changed the value before it was + used in the caller. Patch by myself, tested by ebroad on + #asterisk ........ + +2009-10-06 Leif Madsen <lmadsen@digium.com> + + * Released Asterisk 1.6.2.0-rc3 + +2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, + channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, + res/res_clialiases.c, /, channels/chan_sip.c, + funcs/func_dialgroup.c, include/asterisk/astobj2.h, + res/res_phoneprov.c: Merged revisions 222176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct + 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 + Oct 2009) | 20 lines Fix ao2_iterator API to hold references to + containers being iterated. See Mantis issue for details of what + prompted this change. Additional notes: This patch changes the + ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum + instead of a macro, with a name that fits our naming policy; + also, it is now necessary to call ao2_iterator_destroy() on any + iterator that has been created. Currently this only releases the + reference to the container being iterated, but in the future this + could also release other resources used by the iterator, if the + iterator implementation changes to use additional resources. + (closes issue #15987) Reported by: kpfleming Review: + https://reviewboard.asterisk.org/r/383/ ........ ................ + + * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c, + configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 + Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be + supportable via configuration option. Many T.38 endpoints + incorrectly send the maximum IFP frame size they can accept as + the T38FaxMaxDatagram value in their SDP, when in fact this value + is supposed to be the maximum UDPTL payload size (datagram size) + they can accept. If the value they supply is small enough (a + commonly supplied value is '72'), T.38 UDPTL transmissions will + likely fail completely because the UDPTL packets will not have + enough room for a primary IFP frame and the redundancy used for + error correction. If this occurs, the Asterisk UDPTL stack will + emit log messages warning that data loss may occur, and that the + value may need to be overridden. This patch extends the + 't38pt_udptl' configuration option in sip.conf to allow the + administrator to override the value supplied by the remote + endpoint and supply a value that allows T.38 FAX transmissions to + be successful with that endpoint. In addition, in any SIP call + where the override takes effect, a debug message will be printed + to that effect. This patch also removes the T38FaxMaxDatagram + configuration option from udptl.conf.sample, since it has not + actually had any effect for a number of releases. In addition, + this patch cleans up the T.38 documentation in sip.conf.sample + (which incorrectly documented that T.38 support was passthrough + only). (issue #15586) Reported by: globalnetinc ........ + +2009-10-02 17:35 +0000 [r222032] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 + (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 + Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a + memcpy. ........ ................ + +2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher <tlesher@digium.com> + + * main/astobj2.c, /: Merged revisions 221971 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) + | 9 lines Merged revisions 221970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) + | 2 lines Ensure the result of the hash function is positive. + Negative array offsets suck. ........ ................ + + * /, main/logger.c: Merged revisions 221920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | + tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines + Initialize a variable that we check immediately upon startup. + (closes issue #15973) Reported by: atis ........ + +2009-10-02 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: + Merged revisions 221844 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) + | 33 lines Merged revisions 221769 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) + | 26 lines Occasionally losing use of B channels in chan_misdn. I + have not been able to reproduce the problem of losing channels. + However, I have seen in the code a reentrancy problem that might + give these symptoms. The reentrancy patch does several things: 1) + Guards B channel and B channel structure allocation. 2) Makes the + B channel structure find routines more precise in locating + records. 3) Never leave a B channel allocated if we received + cause 44. The last item may cause temporary outgoing call + problems, but they should clear when the line becomes idle. + (closes issue #15490) Reported by: slutec18 Patches: + issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett + (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) + Reported by: FabienToune Patches: + issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett + (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ + ................ + +2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions + 221777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) + | 9 lines Merged revisions 221776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) + | 2 lines Fix a bunch of off-by-one errors ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | + tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines + Revision 220906 (a merge from 1.4) was not merged correctly, + causing a problem with non-dynamic peers. ........ + +2009-10-01 19:35 +0000 [r221698] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | + dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines + outbound tls connections were not defaulting to port 5061 (closes + issue #15854) Reported by: dvossel Patches: + sip_port_config_trunk.diff uploaded by dvossel (license 671) + Tested by: dvossel ........ + +2009-10-01 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 221554,221589 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, + 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE + constructs when it's just TRUE or FALSE. ................ r221589 + | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 + lines Merged revisions 221588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct + 2009) | 2 lines Use unsigned ints for portinuri flags. ........ + ................ + +2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com> + + * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged + revisions 221592 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | + kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 + lines Remove ability to control T.38 FAX error correction from + udptl.conf. chan_sip has had the ability to control T.38 FAX + error correction mode on a per-peer (or global) basis for a + couple of releases now, which is where it should have been all + along. This patch removes the ability to configure it in + udptl.conf, but issues a warning if the user tries to do, telling + them to look at sip.conf.sample for how to configure it now. For + any SIP peers that are T.38 enabled in sip.conf, there is already + a default for FEC error correction even if the user does not + specify any mode, so this change will not turn off error + correction by default, it will have the same default value that + has been in the udptl.conf sample file. ........ + +2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 | + mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2 + lines Cleaned up merge from r221432 ........ + + * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions + 221432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep + 2009) | 17 lines Merged revisions 221360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep + 2009) | 10 lines Fix SRV lookup and Request-URI generation in + chan_sip. This patch adds a new field "portinuri" to the sip + dialog struct and the sip peer struct. That field is used during + RURI generation to determine if the port should be included in + the RURI. It is also used in some places to determine if an SRV + lookup should occur. (closes issue #14418) Reported by: klaus3000 + Tested by: klaus3000, mnicholson Review: + https://reviewboard.asterisk.org/r/369/ ........ ................ + +2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com> + + * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | + mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines + Prevents from division by zero ........ + + * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged + revisions 221368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | + 23 lines Merged revisions 221153,221157,221303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | + 2 lines check bounds - prevents for buffer overflow ........ + r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | + 8 lines added a new dialplan function 'CSV_QUOTE' and changed the + cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr + Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: + mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, + 30 Sep 2009) | 2 lines changed the prototype definition of + csv_quote ........ ................ + +2009-09-30 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com> + + * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c, + include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) + | 32 lines Merged revisions 221086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) + | 25 lines Change the SSRC by default when our media stream + changes Be default, change SSRC when doing an audio stream + changes Asterisk doesn't honor marker bit when reinvited to + already-bridged RTP streams,resulting in far-end stack discarding + packets with "old" timestamps that areactually part of a new + stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is + a reinvite, unless the 'constantssrc' is set to true in sip.conf. + The original issue reported to Digium support detailed the + following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based + Application Server Call comes in fromITSP, Asterisk dials the app + server which sends a re-invite back toAsterisk--not to negotiate + to send media directly to the ITSP, but to indicatethat it's + changing the stream it's sending to Asterisk. The app + servergenerates a new SSRC, sequence numbers, timestamps, and + sets the marker bit on the new stream. Asterisk passes through + the teimstamp of the new stream, butdoes not reset the SSRC, + sequence numbers, or set the marker bit. When the timestamp on + the new stream is older than the timestamp on the originalstream, + the ITSP (which doesn't know there has been any change) discards + the newframes because it thinks they are too old. This patch + addresses this by changing the SSRC on a stream update unless + constantssrc=true is set in sip.conf. Review: + https://reviewboard.asterisk.org/r/374/ ........ ................ + +2009-09-30 16:57 +0000 [r221204] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 221201 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) + | 14 lines Merged revisions 221200 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) + | 7 lines Avoid a potential NULL dereference. (closes issue + #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt + uploaded by tilghman (license 14) Tested by: kobaz ........ + ................ + +2009-09-30 14:57 +0000 [r221089] Sean Bright <sean@malleable.com> + + * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep + 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() + option. We require box numbers, not names as the documentation + implies. (issue #14740) Reported by: pj Patches: + __20090729-app_voicemail-documentation.patch uploaded by lmadsen + (license 10) Tested by: seanbright, lmadsen ........ + +2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_lock.c: Recorded merge of revisions 221044 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29 + Sep 2009) | 8 lines Allow locks to be inherited through a + masquerade without causing starvation. (closes issue #14859) + Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded + by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt + uploaded by tilghman (license 14) Tested by: atis, tilghman + ........ + + * include/asterisk/smdi.h, include/asterisk/optional_api.h + (removed), apps/app_voicemail.c, include/asterisk/agi.h, + include/asterisk/monitor.h: Remove optional_api from 1.6.2 + branch, since it is not currently working. This is a blocking + issue for the 1.6.2 release. (closes issue #15914) Reported by: + mbeckwell Branch: + http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 + Tested by: mbeckwell + + * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) + | 16 lines Merged revisions 220873 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) + | 9 lines Reduce CPU usage related to building a peer merely for + devicestates. This fixes a 100% CPU problem in the SIP driver, + found by profiling the driver while the problem was occurring. + (closes issue #14309) Reported by: pkempgen Patches: + 20090924__issue14309.diff.txt uploaded by tilghman (license 14) + Tested by: pkempgen, vrban ........ ................ + +2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the + spyee is masqueraded and chanspy_ds_chan_fixup() is called with + the channel locked. (closes issue #15965) Reported by: atis + Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson + (license 96) Tested by: atis + + * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep + 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the + code. Also document the fact that app_confbridge does not + automatically answer the channel. (closes issue #15964) Reported + by: shrift ........ + +2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) + | 12 lines Make deletion of temporary greetings work properly + with IMAP_STORAGE When imapgreetings was set to yes, the message + was being deleted but wasn't actually being expunged. When + imapgreetings was set to no, the file based message was not being + deleted at all. All good now! (closes issue #14949) Reported by: + noahisaac Patches: vm_tempgreeting_removal.patch uploaded by + noahisaac (license 748), modified by me ........ + +2009-09-28 19:13 +0000 [r220725] Sean Bright <sean@malleable.com> + + * /, Makefile.rules: Merged revisions 220721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep + 2009) | 10 lines Merged revisions 220717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep + 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, + explicitly pass -O0 to the compiler so we override any default + optimization levels for a particular install. ........ + ................ + +2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 | + jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines + Fix building of registration entry in build_peer when using + callbackextension Check for remotesecret option was + unintentionally always true, which therefore caused the secret + option to never be used. Thanks to dvossel for pointing out the + exact fix. (closes issue #15943) Reported by: tpsast ........ + +2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info> + + * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) + | 3 lines add name argument for the CALLERID dialplan function to + the xml documentation. Pointed out to me on IRC by snuff-home. + Thanks ........ + +2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) + | 2 lines Allow AES to compile, when OpenSSL is not present. + ........ + +2009-09-24 20:38 +0000 [r220369] David Vossel <dvossel@digium.com> + + * main/tcptls.c, /: Merged revisions 220365 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 | + dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines + fixes tcptls_session memory leak caused by ref count error + (closes issue #15939) Reported by: dvossel Review: + https://reviewboard.asterisk.org/r/375/ ........ + +2009-09-24 19:42 +0000 [r220292] Tilghman Lesher <tlesher@digium.com> + + * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged + revisions 220289 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) + | 13 lines Merged revisions 220288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) + | 6 lines Implicitly sending a progress signal breaks some + applications. Call Progress() in your dialplan if you explicitly + want progress to be sent. (Reverts change 216430, closes issue + #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing + list + http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html + ........ ................ + +2009-09-24 18:22 +0000 [r220103-220221] Sean Bright <sean@malleable.com> + + * Makefile, /: Merged revisions 220217 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep + 2009) | 9 lines Merged revisions 220213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep + 2009) | 1 line Resolve parallel build warnings. Reported by Klaus + Darilion on the asterisk-dev mailing list. ........ + ................ + + * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400 + (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, + 24 Sep 2009) | 2 lines Remove the remaining bashisms in the + Makefile/mkpkgconfig ........ ................ + +2009-09-24 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info> + + * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 + (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) + | 7 lines mkpkgconfig does not need bash so make it use /bin/sh + This fixes building on all systems that don't have bash at + /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on + #asterisk-dev ........ ................ + +2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com> + + * apps/app_directory.c, /: Merged revisions 219987 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) + | 8 lines Fix two possible crashes, one only in 1.6.1 and one in + 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg + Patches: 20090914__issue15739.diff.txt uploaded by tilghman + (license 14) 20090922__issue15739.diff.txt uploaded by tilghman + (license 14) Tested by: DLNoah, jeffg ........ + +2009-09-22 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 + (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) + | 10 lines When IMAP variables were changed during a reload, + Voicemail did not use the new values. This change introduces a + configuration version variable, which ensures that connections + with the old values are not reused but are allowed to expire + normally. (closes issue #15934) Reported by: viniciusfontes + Patches: 20090922__issue15934.diff.txt uploaded by tilghman + (license 14) Tested by: viniciusfontes ........ ................ + +2009-09-21 17:01 +0000 [r219722] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 + (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 + Sep 2009) | 3 lines Reverting merge 219520. This change was not + necessary. ........ ................ + +2009-09-20 18:21 +0000 [r219669] Tilghman Lesher <tlesher@digium.com> + + * /, main/file.c: Merged revisions 219654 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) + | 15 lines Merged revisions 219653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) + | 8 lines Really stop the stream, when ast_closestream() is + called. (closes issue #15129) Reported by: bmh Patches: + 20090918__issue15129.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/372/ ........ + ................ + +2009-09-19 03:14 +0000 [r219590] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r219587 | russell | 2009-09-18 21:59:52 -0500 + (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) + | 6 lines Make sure the iax_pvt exists before dereferencing it. + This fixes the latest crash posted on issue 15609. (issue #15609) + ........ ................ + +2009-09-18 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 + (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) + | 9 lines iax2 frame double free The iax frame's retrans sched id + was written over right before iax2_frame_free was called. In + iax2_frame_free that retrans id is used to delete the sched item. + By writing over the retrans field before the sched item could be + deleted, it was possible for a retransmit to occur on a freed + frame. ........ ................ + + * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) + | 20 lines Merged revisions 219450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) + | 14 lines via-header branches not updated correctly on INVITE + INVITE requests must always contain a new unique branch id. When + a new branch id is created for an INVITE, the dialog's + invite_branch variable must be updated so CANCEL requests use the + correct branch id. (closes issue #15262) Reported by: maniax + Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety + (license 608) invite_new_branch_trunk.diff uploaded by dvossel + (license 671) Tested by: maniax, dvossel ........ + ................ + +2009-09-18 13:57 +0000 [r219415] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) + | 6 lines Missing value setting line for maxsecs/maxmessage + (closes issue #15696) Reported by: fhackenberger Patches: + maxsecs.patch uploaded by fhackenberger (license 592) ........ + +2009-09-17 22:38 +0000 [r219376] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 | + dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines + fixes deadlock when performing directed pickup w Invite/replaces + (closes issue #15340) Reported by: lmsteffan Patches: + deadlock.patch uploaded by lmsteffan (license 779) Tested by: + lmsteffan ........ + +2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep + 2009) | 12 lines Merged revisions 219320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep + 2009) | 6 lines Send a 100 Trying response when we detect a + spiral. This was problematic during spiral tests at SIPit... + along with some other things as well. ........ ................ + +2009-09-17 22:06 +0000 [r219307] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) + | 27 lines Merged revisions 219303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) + | 21 lines INVITE w/Replaces deadlock fix This patch cleans up + the locking logic in chan_sip.c's handle_invite_replaces() + function as well as making use of ast_do_masquerade() rather than + forcing the masquerade on an ast_read(). The code had several + redundant unlocks that would result in 'freed more times than + we've locked!' errors. I cleaned these up as well as moving all + the unlock logic to the end of the function. This patch should + also resolve the issue people were having with the replacecall + channel never being unlocked with one legged calls. (closes issue + #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff + uploaded by dvossel (license 671) Tested by: irroot, dvossel + Review: https://reviewboard.asterisk.org/r/371/ ........ + ................ + +2009-09-17 19:58 +0000 [r219267] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | + file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines + Ensure no spaces exist before "refresher=" when doing the + comparison. ........ + +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 vi |