diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-02 23:16:10 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-03-02 23:16:10 +0000 |
| commit | 1bc04a40a2f44ac8a358802e316af8afac8afb8f (patch) | |
| tree | 5b8917198c3c43547a3f7da60cf93c57fd3c4030 /ChangeLog | |
| parent | c3644444105a1e15dbaa2ca69eee2169aa48b6fb (diff) | |
importing files for 1.4.1 release
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.1@57561 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 5181 |
1 files changed, 5181 insertions, 0 deletions
diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..1b95a1770 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,5181 @@ +2006-03-02 Russell Bryant <russell@digium.com> + + * Asterisk 1.4.1 released. + +2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com> + + * configure, configure.ac: Update the check that is used to + determine whether zaptel transcoder support is present. The + interface has changed. + +2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 + lines If a SIP message comes in and goes to a method handler that + requires additional values that may not be present then send back + an error. ........ + +2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 57458 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 + line further refinement in wording of goto documentation, as per + 9156, goto not proceeding to next instruction ........ + + * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes + right, but 9184 points out the problem-- the escape is removed by + pbx_config, and pbx_ael should also, before sending it down into + the pbx engine. Also, you have to insert it back in, if you are + generating extensions.conf code from the AEL. + +2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com> + + * main/file.c: Return the correct digit that interrupted the + stream. This fixes exiting the Background application when using + the m option. (issue #9176, mjagdis) + + * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, + include/asterisk/channel.h: Merge changes from + svn/asterisk/team/russell/sla_updates * Originally, I put in the + documentation that only Zap interfaces would be supported on the + trunk side. However, after a discussion with Qwell, we came up + with a way to make IP trunks work as well, using some things + already in Asterisk. So, here it is, this now officially supports + IP trunks. * Update the SLA documentation to reflect how to setup + IP trunks. * Add a section in sla.txt that describes how to set + up an SLA system with voicemail. * Simplify the way DTMF + passthrough is handled in MeetMe. * Fix a bug that exposed itself + when using a Local channel on the trunk side in SLA. The + station's channel needs to be passed to the dial API when dialing + the trunk. * Change a WARNING message to DEBUG in channel.h. This + message is of no use to users. + +2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, /: Merged revisions 57317 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar + 2007) | 2 lines Don't even attempt to optimize things when a + proxy channel is involved. It will just explode in weird and + unexplaineable ways. (issue #9175 reported by + clegall_proformatique) ........ + +2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com> + + * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. + +2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com> + + * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla + docs + + * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes + from svn/asterisk/team/russell/sla_updates * Add support for + private hold. By setting "hold=private" for a trunk, only the + station that put the call on hold will be able to retrieve it + from hold. Also, by setting "hold=private" for a station, any + call that station puts on hold can only be retrieved by that + station. + + * apps/app_meetme.c: Minor formatting change + + * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from + svn/asterisk/team/russell/sla_updates * Add support for the + "barge=no" option for trunks. If this option is set, then + stations will not be able to join in on a call that is on + progress on this trunk. + +2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 57118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 + line a small documentation update, to reflect reality in the goto + doc strings, as per 9156, Goto does not proceed to next prio if + jump fails ........ + +2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb + 2007) | 2 lines Fix a few more issues with the agent logoff CLI + command. (issue #9123 reported by arbrandes) ........ + +2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com> + + * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of + changes from svn/asterisk/team/russell/sla_updates * Add support + for station ring delays. Ring delays can be set globally for a + station or for specific trunks on the station. * Fix a few bugs + in existing code. * Restructure and Reorganize code to improve + readability and maintainability. * Improve formatting of the "sla + show (trunks|stations)" CLI commands. + +2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com> + + * apps/app_meetme.c: Picky compiler... + + * apps/app_speech_utils.c: Better handle timeouts when the + individual speaks after everything has been played but before the + timeout ends. + +2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: I was surprised that I had not yet downgraded + missing goto targets and macro call defs to a warning, in case + they are in extensions.conf; I rectified this problem. Also, A + goto in a macro to a target in a catch block was not being found; + I fixed this too; the cause was that I needed to treat catch + statements like an extension in the find_match code. + +2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: Fix voicemail email attachments. I missed + the conversion of one of the line endings and there was an extra + one where it should not have been. (issue #9128) + +2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, + picky... show deprecation warning in application help, too + (reported via list) + +2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com> + + * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where + if a device was not specified in alsa.conf, then we just use the + system default, instead of creating our own default of hw:0,0. + (issue #9139) + +2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com> + + * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 + lines Obey the clearglobalvars option in extensions reload (or + dialplan reload depending on your version). (issue #9146 reported + by ramonpeek) ........ + +2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Fix a crash in my last change to + iax2_indicate(). (issue #9150) + +2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com> + + * apps/app_record.c: Update app_record documentation to use new CLI + command, core show file formats. (issue #9151 reported by junky) + + * main/pbx.c: Use ast_strlen_zero to see if the language and/or + context argument is not present for Background instead of just + checking if it is NULL. (issue #9141 reported by mjagdis) + +2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Do more complete locking of the + chan_iax2_pvt struct in the indicate callback. (Problem brought + up by Ben Smithurst on the asterisk-dev list) + +2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com> + + * main/asterisk.c: Allow both of the show version files and core + show file versions CLI commands to work. (issue #9135 reported by + mvanbaak) + +2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Move a comment to be in the correct struct. + + * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell + | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure + that lock.h is included in utils.c with AST_API_MODULE defined so + that the implementations will be properly included when the + AST_INLINE_API functions are not going to be inlined. (issue + #9124, festr) ........ + +2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/channel.c, /: Merged revisions 56684 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) + | 3 lines Issue 9130 - If prev is the last item on the channel + list, then evaluating additional conditions (e.g. name prefix) + will cause a NULL dereference. ........ + +2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Make sure to set a speeddials parent on + creation. Don't crash if hold is pressed when no call is active. + Don't return in places that we shouldn't.. + +2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com> + + * codecs/codec_zap.c: update to match zaptel 1.4 API change that + was committed a few minutes ago + +2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 56504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | + 8 lines Fix up a couple more signal handlers to not do bad things + that could cause various undesirable results. The other day, I + made Asterisk deadlock by hitting Control-C because of a bad + signal handler. Now, signal handlers just set a flag and write to + an alert pipe for the flag to be handled. Then, there is another + thread that is monitoring for these flags. If being run in + console mode, it is just the main thread. If Asterisk is in the + background, a thread is created to do it. ........ + +2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com> + + * main/sched.c: Change log notice to debug. It is possible for a + scheduled item to execute and be deleted at close to the same + time and unavoidable. If this happens this message creeps up. + +2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | + 4 lines Don't destroy mutexes before unregistering all of the + entry points from the core. Also, fix a potential memory leak + from not destroying the locks for all of the possible call + numbers (about 32k of them). ........ + +2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com> + + * build_tools/make_version_h: build special version strings for + AADK/S800i builds + +2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: The IMAP storage code uses the same code to + build the email that is used when voicemail is sent via email + using something like sendmail. In the patch from bug 8033 to fix + various IMAP storage problems, the line endings in the email file + were changed in the code from "\n" to "\r\n". However, this + breaks sending regular voicemail to email. So, this change + conditionally sets line endings to "\r\n" only if IMAP_STORAGE is + enabled. (issue #9128, patch by jarjarbinks, modified by me to + not break IMAP storage) + +2007-02-22 23:25 +0000 [r56280] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 56279 via svnmerge ........ r56279 | file | + 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always + defer Agent logoff if any channels are up until they hang up. + (issue #9123 reported by arbrandes) ........ + +2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com> + + * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, + doc/sla.txt: Merge changes from team/russell/sla_updates. This + batch of changes to the SLA code does a few different things. * I + made the SLA code event driven instead of having to act in a lot + of busy loops while dialing things to wait for state changes. + This makes the code more efficient and readable at the same time. + * I have implemented a couple of new features. The first is + inbound trunk ringing timeouts. This is an option that defines + how long to let an incoming call on a trunk to ring. * I have + also implemented ring timeouts for stations. They may be + specified for the entire station, meaning it is how long to let + the station ring before giving up. You can also specify a ring + timeout for a specific trunk on a station. So, you can say that + you only want a specific station to ring 5 seconds if it is line1 + ringing, but otherwise, there is no timeout. + +2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 + lines Only change the original or clone channel if it's the + channel behind the proxy channel, not if it's just a regular + bridged channel. ........ + +2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com> + + * doc/osp.txt: Update OSP documentation for v1.4. + +2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Move message from verbose to debug + +2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com> + + * sounds/Makefile: updated the sound tarball versions in Makefile + +2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Restructure a little bit of code to reduce + nesting. There is no functionality change here. + + * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | + 3 lines If we receive a frame that is not in any of the + negotiated formats, then drop it. (potentially issue #8781 and + SPD-12) ........ + +2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com> + + * main/cli.c: Print out deprecation notice on usage output of CLI + commands. (issue #8925 reported by blitzrage) + +2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com> + + * main/loader.c: disable unloading of embedded modules... there is + a fundamental problem with doing so that will not be fixed in + this version of Asterisk due to its invasiveness + +2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 + lines Change naughty warning message to provide useful + information. If a write now fails on a channel in meetme it will + tell you the channel name instead of spitting out the wrong error + message. ........ + +2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com> + + * channels/chan_gtalk.c: Fix locking issue, and accept + "transport-accept" as a valid accept message. This should solve + issues 8970 and 8503. + +2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Simplify the last change to app_meetme, and + move the call to dispose_conf() up into the block where we know a + conf exists. + +2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com> + + * apps/app_meetme.c: Only dispose of the conference if one was + created. + + * apps/app_speech_utils.c: Only start playing the next file if we + have not been quieted. + + * channels/chan_sip.c: Add a flag that indicates whether a SIP + dialog is an outgoing call or not. SIP_OUTGOING originally did it + but it was repurposed to the direction of the last transaction, + which can cause update_call_counter to falsely decrease the wrong + counters. (please don't hurt me oej) (issue #8943 reported by + mdu113) + +2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com> + + * /, build_tools/make_version: Merged revisions 55868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 + Feb 2007) | 2 lines use new tag version script ........ + +2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly + after transfer (decrement inuse early on transferer's call leg) + +2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com> + + * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. + Issue 7764, patch by sailer + +2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Improve the reference counting to fix bugs + where people report seeing conferences listed that have no + members. (issue #9073) + + * /: Blocked revisions 55750 via svnmerge ........ r55750 | russell + | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix + random crashes when using the MeetMe application. This patch + converts list handling to use the linked list macros and most + importantly, implements reference counting on the ast_conference + objects. The reference counting was first backported from 1.4. + However, that code has some problems that caused the reference + count to never hit zero. Those problems are fixed in this patch + and will be resolved in 1.4 and trunk next, with a different + patch. (issues #7647, #9073, #9106, BE-115). ........ + +2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Better handle dropped IMAP connections. + (issue #9054 reported by bsmithurst) + + * channels/chan_sip.c: Return behavior I removed. I did not + remember that you could just add a localnet entry to make it + work. + + * channels/chan_sip.c: Don't test our own address against the + localnet settings. At least one person has had issues as a result + of this from #7051 so I'm reversing it. (issue #8821 reported by + kokoskarokoska) + + * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb + 2007) | 2 lines Defer clearing callback information if channels + are up until they are hung up. This ensures the hangup process + goes smoothly and no channels get hung in limbo. (issue #8088 + reported by kebl0155) ........ + +2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com> + + * main/http.c: Add the Asterisk version information to the Server + header in HTTP responses. (requested by Pari) + + * include/asterisk/manager.h: Increase the maximum number of + manager headers to 128, at the request of Pari. + + * /: Blocked revisions 55588 via svnmerge ........ r55588 | russell + | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert + a tab to spaces so that the documentation is printed out properly + aligned. ........ + +2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com> + + * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free + with strdupa (thanks file) 55555! + +2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com> + + * configs/sla.conf.sample: Change the formatting of sla.conf.sample + to make it more readable. (issue #9112, blitzrage) + +2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net> + + * res/res_jabber.c: - Not sending arguments to an application is + not "out of memory" - Making error messages a bit more clear + +2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) + | 2 lines forcename and forcegreetings options should check to + see if the recording already exists ........ + +2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com> + + * channels/chan_iax2.c: Changed iax2 process thread to detached to + correct memory leak due to left over thread context on thread + exit. Modified module unload process to avoid deadlocks on + pthread cancels + +2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net> + + * /, apps/app_record.c: Merged revisions 55277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 + lines Documentation update (#9053, jsmith) ........ + + * /: Block patch that was made only for 1.2 (already implemented in + 1.4 and trunk) + +2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com> + + * apps/app_queue.c: Add missing membername option to AddQueueMember + documentation. (issue #9088 reported by seanbright) + +2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Fix an issue where callerid would not be + displayed on some phones. Issue 8995, initial patch and research + done by wedhorn + +2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 + lines Answer the channel before recording privacy information. + (issue #8926 reported by lmamane) ........ + + * apps/app_queue.c: Make the 'i' option of Queue actually work. + (issue #8986 reported by utis) + + * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 + lines Allow chan_sip to handle attended transfers from a SIP + phone that is sitting behind chan_agent. Yes folks, all it took + was one line of code. (issue #8784 reported by pzieba) ........ + +2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac: If the + pg_config application is found, but there is probably executing + it, then consider postgres unavailable. (issue #8637) + + * codecs/gsm/Makefile: Filter out yet another architecture that + does not work with the optimizations in the built-in libgsm. + (issue 8637, ovi) + + * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged + revisions 55005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | + 9 lines Revert the change I did in revisions 54955, 54969, and + 54970, in 1.2, 1.4, and trunk. I decided that once a conference + is created from meetme.conf, it is acceptable behavior that the + pin can not be changed until the conference goes away. I also + added a note in meetme.conf to describe this behavior. We still + have another issue in 1.4 and trunk where some conferences with + no users don't go away. That is the real bug that needs to be + addressed here. ........ + +2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb + 2007) | 2 lines Do not send indications through ast_indicate in + chan_agent but instead go directly to the technology. This way + when indications are emulated they happen on the Agent channel + and do not screw up formats on the channels. (issue #8439 + reported by punkgode) ........ + +2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | + 5 lines For conferences that are configured in meetme.conf, check + the configuration file every time someone joins the conference + instead of only when the conference is first created. This is to + ensure that changes to the pin numbers in the config file are + always honored. (issue #9073) ........ + +2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: Need to check macro extension as well as macro + context for directed pickup. + +2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com> + + * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by + default. It was set to enabled in pbx.c. However, if the option + was not present in extensions.conf, then pbx_config.c would set + it back to disabled. + + * res/res_features.c: Clean up a few coding guidelines issues - + spaces to tabs, use sizeof() to pass the size of a static buffer, + add spaces ... + +2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com> + + * main/asterisk.c: Clarify a restart message. It's silly, but the + reporter had a very valid point. Issue 9079 + +2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: Allow directed pickup to pick up the real + context instead of the macro context if a Macro is used. (issue + #8984 reported by jamesb63) + +2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue #7541 - Handle multipart attachments + to SIP messages - even if boundary is quoted. + + * /, res/res_agi.c: Merged revisions 54771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 + lines Issue #9069 - If we open with TH we should not close with + /TD. (seanbright) ........ + +2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com> + + * apps/app_speech_utils.c: Don't let dtmf leak over into the engine + and let it skew the results... also give DTMF results priority. + (issue #9014 reported by surftek) + + * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 + lines Use a separate variable to indicate execution should + continue instead of the return value. (issue #8842 reported by + pluto70) ........ + + * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue + #9068 reported by mhardeman) + +2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net> + + * /: Block patch only needed in 1.2 + +2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com> + + * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 + lines When handling glare on a PRI, move the requested channel + rather than hang up the old one. Fix for 8957 and 9011. ........ + +2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees + with it's placement, feel free to change it. (issue #9045 + reported by gork) + +2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Remove a couple of leftover debug messages + + * include/asterisk/devicestate.h: Fix the documentation on the + return values from device state provider registration and + deletion. + + * channels/chan_sip.c: If we fail to create the SIP socket, then + return -1 from reload_config() so that load_module() will return + AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get + spammed with error messages every time chan_sip tries to send a + message. + +2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net> + + * /: Blocking patch for 1.2 only + +2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com> + + * main/dial.c, include/asterisk/dial.h: Change + ast_set_state_callback() to ast_dial_set_state_callback() + + * main/dial.c, apps/app_meetme.c, apps/app_page.c, + include/asterisk/dial.h: - Add the ability to register a callback + to monitor state changes in an asynchronous dial operation. - + Rename the various references to "status" to "state" in the dial + API + +2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com> + + * configure, configure.ac: Make the --without-oss argument work. + (issue #9026 reported by puzzled) + +2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com> + + * configs/users.conf.sample: Fix a typo where "vmpassword" should + be "vmsecret" + +2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_h323.c: Fix VLDTMF reception + + * apps/app_echo.c: Much simpler than previous one ;-) + + * main/channel.c: Provide correct DTMF duration + + * main/cli.c: Bring deprecated 'debug channel <x|all>' command back + +2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com> + + * configure, configure.ac, acinclude.m4: don't display the + --with-imap message unless --with-imap was specified without a + path use '-n' instead of '! -z' for tests + +2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Add some output for "show application + SLAStation/SLATrunk" + + * channels/chan_sip.c: Change some text to properly state "On + Hold", which was already done in trunk. + + * configs/sla.conf.sample, include/asterisk/app.h, + include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, + channels/chan_sip.c, doc/sla.txt (added), + include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge + team/russell/sla_rewrite This is a completely new implementation + of the SLA functionality introduced in Asterisk 1.4. It is now + functional and ready for testing. However, I will be adding some + additional features over the next week, as well. For information + on how to set this up, see configs/sla.conf.sample and + doc/sla.txt. In addition to the changes in app_meetme.c for the + SLA implementation itself, this merge brings in various other + changes: chan_sip: - Add the ability to indicate HOLD state in + NOTIFY messages. - Queue HOLD and UNHOLD control frames even if + the channel is not bridged to another channel. linkedlists.h: - + Add support for rwlock based linked lists. dial.c: - Add the + ability to run ast_dial_start() without a reference channel to + inherit information from. + + * apps/app_echo.c: When the Echo() application receives the digit + '#', echo that back as well. Since we already sent the BEGIN + frame for that digit, it makes sense to send the END as well. + +2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_gtalk.c: another dependency + + * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, + funcs/func_odbc.c, res/res_adsi.c: add some inter-module + dependencies + + * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk + scripts to work when both MODULEINFO and MAKEOPTS are present in + a source file + +2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: Temporarily change musicclass on channel to one + specified in Dial so that the 'm' option functions properly. + (issue #8969 reported by christianbee) + +2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com> + + * doc/imapstorage.txt, configure, configure.ac: clarify the fact + that voicemail IMAP storage cannot be built against a distro's + binary c-client library package (at least not at this time) + +2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net> + + * main/acl.c: Don't output debug unless we asked for it + +2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com> + + * apps/app_speech_utils.c: Fix timeout issue when utterance is + longer then timeout itself. + +2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/loader.c: Issue 9007 - Mutex not released on early return + + * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) + | 2 lines Issue 9003 - If fullname is empty, quote() passes back + "\"" ........ + +2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com> + + * main/db1-ast/Makefile: When building libdb1.a, put the additional + flags needed at the beginning of ASTCFLAGS, instead of at the + end. This way, we ensure that we find the local headers first + before accidentally trying to use headers that exist in locations + specified in the ASTCFLAGS passed from the main Makefile. (issue + #8637, ovi) + + * main/Makefile: The clean target actually needs to run "distclean" + on editline. This is because we need to make sure that its + configure script gets executed again, because the CFLAGS we want + to pass to editline may have changed. + +2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: We can not reliably do P2P bridging with DTMF passing + back with compensation if we need to listen for DTMF frames. + (issue #8962 reported by caio1982) + +2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com> + + * main/rtp.c: When parsing the NTP timestamp in a sender report + message, you are supposed to take the low 16 bits of the integer + part, and the high 16 bits of the fractional part. However, the + code here was erroneously taking the low 16 bits of the + fractional part. It then shifted the result 16 bits down, so the + result was always zero. This fix makes it grab the appropriate + high 16 bits, instead. (issue #8991, pointed out by + andre_abrantes) + +2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com> + + * apps/app_playback.c: Directly load say.conf in load_module + instead of calling the reload function. (issue #8946 reported by + junky) + + * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 + lines Fix a few potential memory leaks with realtime users and + peers. (issue #8999 reported by bsmithurst) ........ + +2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) + | 2 lines Issue 7440 - Macro called from Macro from the h + extension exits prematurely ........ + +2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 52843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | + 1 line fixed some possible segfaults. also fixed an very + important bug which occurs on high load (when calls are very fast + generated) ........ + +2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * res/res_jabber.c: Text fix for jabber reload command (reported by + bkruse via IRC) + + * main/manager.c, /: Merged revisions 53245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) + | 2 lines Issue 8987 - Status could return two responses + (mnicholson) ........ + +2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Formatting + +2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com> + + * apps/app_playback.c: Ensure say_cfg is NULL when the module is + loaded. (issue #8946 reported by junky) + + * apps/app_playback.c: Unregister Playback CLI commands as well as + dialplan application. (issue #8946 reported by junky) + +2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Add some comments on queue system behaviour + and how it affects the SIP channel + +2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Make SIPDtmfMode application work with + recent capability changes, and also fix an RTP stack issue when + the auto option was used. (issue #8972 reported by mdu113) + +2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com> + + * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | + 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when + the dial application exits early because of invalid arguments + instead of just leaving it empty. (issue #8975) ........ + + * /: Blocked revisions 53134 via svnmerge ........ r53134 | russell + | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert + some changes that accidentally got committed as a part of another + fix. ........ + +2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string + because due to compatibilities with CS1000 reported at + www.voip-info.org + +2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com> + + * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a + warning to the console that things might possibly be + misconfigured when queue member's states are still 'Not in Use' + when we're about to bridge them with a caller from queue. Also, + put some documentation quoted from oej's queues.txt efforts + started in /trunk today. This commit puts #7433 into feedback + state for 1.4, and pending no further negative feedback, it will + finally be closed. + +2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Correct a copy/pasted error message line for RTCP. + + * main/config.c, /: Merged revisions 53117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 + lines Pass the glob expanded filename to process_text_line so + that error messages contain the actual filename, not the original + include one. (issue #8959 reported by tzafrir) ........ + + * Makefile: Add systemname to asterisk.conf generation per recent + discussions about it. (issue #8968 reported by blitzrage) + +2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct + p2p RTP call setup in SIP. You can enable it in sip.conf, but it + is now considered experimental until we solve the + AST_CONTROL_ANSWER with payload and videocaps stuff. + +2007-02-01 23:16 +0000 [r53108] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell | + 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a + small typo. Synopsis lines shouldn't have a newline ........ + +2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 + lines Copy noncodeccapability over to the joint variable so that + telephone-event will get transmitted in the sent INVITE. ........ + + * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile + here as well, but it apparently required both dev mode and no + optimizations to creep up. + + * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 + lines Don't negotiate RFC2833 when not configured to do so. + (issue #8799 reported by mdu113) ........ + +2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com> + + * funcs/func_strings.c: Fix the FIELDQTY function to not crash. + (reported by blitzrage and Corydon on IRC) + +2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net> + + * /: Going backwards, blame file. + +2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb + 2007) | 2 lines Return previous behavior of having MOH pick up + where it was left off. (issue #8672 reported by sinistermidget) + ........ + + * funcs/func_strings.c: Make func_strings build under dev mode. + Didn't I do this today already in the berkeley DB? + +2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement + call counter - If it's still set at time of dialog destruction, + make sure we decrement the device call counter properly before we + destroy the dialog + + * apps/app_queue.c: Change debug level for state change message + that is not really informative when debugging app_queue + + * channels/chan_sip.c: Cleaning up the devicestate callback + function + +2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * funcs/func_strings.c: Oops. + + * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) + | 2 lines Bug 8965 ........ + +2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com> + + * main/asterisk.c: Add missing 'F' letter to getopt so it magically + becomes a valid option. (issue #8960 reported by tzafrir) + +2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) + | 2 lines No wonder FIELDQTY doesn't work with functions... the + documentation in pbx.c was wrong ........ + +2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix silly logic. We really want to write + UDPTL frames out when the call is up. + +2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Add explanation of port= in combination + with defaultip= (thanks jsmith) + +2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: we update the name on any first reply of + our setup + +2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_h323.c: chan_h323 is very stable, so let it built + by default + +2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: When going on hold have the side that was put on hold + reinvite back to Asterisk. When going off hold have the side that + was taken off hold reinvited back to the other party. + + * main/rtp.c: Add more frame types to forward in the RTP bridge + loops. + +2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com> + + * main/cdr.c, main/manager.c, pbx/pbx_spool.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, + channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: + Merged revisions 53045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | + 3 lines Fix a bunch of places where pthread_attr_init() was + called, but pthread_attr_destroy() was not. ........ + + * apps/app_userevent.c: Remove an extra \r\n from manager user + events. (issue #8955, mnicholson) + + * main/rtp.c, /: Merged revisions 53039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | + 3 lines Use the proper format string to print unsigned values in + the rtp debug output. (issue #8954, wmis) ........ + + * apps/app_queue.c: Only changed the paused status in an existing + queue member if the paused column exists. + + * apps/app_queue.c: Instead of always creating a realtime queue + member as unpaused, read the "paused" column and use that value + for the paused status of the member. (issue #8949, jmls) + + * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. + (issue #8363, johnlange) + + * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue + #8942, lters) + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + codecs/codec_gsm.c: When we are checking for a system installed + version of libgsm, we need to check for gsm.h as well. + Furthermore, when checking for this header, it may be located in + a gsm/ sub directory, so check for that, as well. (issue #8773) + + * /: Blocked revisions 52954 via svnmerge ........ r52954 | russell + | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't + print a message indicating that we don't know what to do with a + proceeding control frame in ast_request_and_dial(). We just need + to ignore it. (reported by JerJer on #asterisk-dev) ........ + + * channels/chan_sip.c: Only set the DTMF flag on the rtp structure + if the DTMF mode is actually RFC2833, not just that it is not + INFO. This makes it get set for inband DTMF as well, which is not + valid. (issue #8936) + + * main/asterisk.c, /: Merged revisions 52903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | + 9 lines The SIGHUP handler was implemented to allow admins to + send SIGHUP to a running Asterisk process to reload the + configuration. However, doing the actual reload in the signal + handler itself is a very bad thing to do, because the reload + process includes calling non-reentrant functions such as + malloc/calloc/etc. If Asterisk is running in the background, then + the reload will happen immediately. However, if running in + console mode, the reload doesn't work until something is typed at + the console. That sort of defeats the purpose, but I don't see an + easy way to get around it at this point. ........ + + * /: Blocked revisions 52857 via svnmerge ........ r52857 | russell + | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment + out the parts in the Makefile that make codec_zap get built. It + will not yet build against zaptel 1.2, so I am disabling it to + prevent further bug reports until it gets merged. (issue #8940) + ........ + +2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Drop the deprecated show commands since the + original ones were changed back. (issue #8937 reported by + PCadach) + +2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_h323.c: Revert reprecation of h.323 gk cycle + command from pre-1.4 version instead of duplicated h323 cycle gk + + * res/res_odbc.c: Don't play with free()'d pointers + + * configure, acinclude.m4: Handle non-standard OpenH323/PWLib + library names + +2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | + 5 lines Fix the extraction of the timestamp from video frames. It + was using the mapping for a mini-frame instead of a video-frame, + which caused it to get invalid data. (issue #8795, mihai) + ........ + +2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com> + + * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan + 2007) | 2 lines Now that filename is part of the structure and + since it comes before postprocess... we have to add it to our + postprocess line. (reported on asterisk-dev by Boris Bakchiev) + ........ + +2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com> + + * main/Makefile: Add a missing quotation mark. This was pointed out + by jcmoore on #asterisk-dev. + + * main/manager.c: Remove a recursive lock of the manager session. + This was pointed out by zandbelt in issue #8711. + +2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * pbx/pbx_config.c: Argument number correction + +2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com> + + * main/Makefile: ASTLDFLAGS needs to be passed to the editline + configure script as LDFLAGS. (issue #8928, zandbelt) + + * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF + mode translation. P2P bridging can only be used when the DTMF + modes don't match if the core is monitoring DTMF in both + directions. Then, the core will handle the translation. + Otherwise, this bridging method can not be used. (issue #8936) + + * main/manager.c: The session lock can not be held while calling + action callbacks. If so, then when the WaitEvent callback gets + called, then no event can happen because the session can't be + locked by another thread. Also, the session needs to be locked in + the HTTP callback when it reads out the output string. This fixes + the deadlock reported in both 8711 and 8934. Regarding issue + 8711, there still may be an issue. If there is a second action + requested before the processing of the first action is finished, + there could still be some corruption of the output string buffer + used to build the result. (issue #8711, #8934) + +2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Use ast_calloc instead of malloc. + +2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com> + + * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR + backport to 1.4). It was committed to trunk via 7663. But it + wasn't so much an enhancement as a fix for the bad language + output for portuguese in Brazil, so, after a lot of prodding from + patient Brazilians, here is the same fix for 1.4 + +2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Set quota information to 0 when creating a + vm_state. (issue #8924 reported by neutrino88) + +2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com> + + * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in + the last commit to the adaptive jitterbuffer code. - Specifically + indicate to the compiler that the "dropem" variable only needs + one but. - Change formatting to conform to coding guidelines. + +2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com> + + * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with + jitterbuf, whereas it would not complain about, and would allow + itself to be overfilled (per the max_jitterbuf parameter). Now it + rejects any data over and above that size, and complains about + it. + +2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * configure, configure.ac: Suggested change to fix normal usage of + --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing + list) + +2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 + lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log + follow documentation. (issue #7677 reported by amilcar) ........ + + * main/manager.c: Have the manager interface send back an "Already + logged in" message instead of "Invalid/Unknown Command" when the + client authenticates for a second time. (issue #8509 reported by + pari) + + * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 + lines Make the last context entry read in the dominant one. + (issue #8918 reported by pj) ........ + + * main/file.c: Fix core show file formats CLI command. + +2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com> + + * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 + lines Allow dequeueing of frames with negative timestamp by + moving jitterbuffer frames check to jb_next. (issue #8546 + reported by harmen) ........ + + * channels/chan_sip.c: Drop out variables I accidentally put in. + + * channels/chan_sip.c: Decrement onHold count if we are hung up on + and still on hold. (issue #8909 reported by alexh42) + + * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan + 2007) | 2 lines Add another note about audio files being played + back to each bridged party. (issue #8718 reported by ppyy) + ........ + +2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c, configs/users.conf.sample: By suggestion + from kpfleming last week, change "vmpassword" to "vmsecret". + + * configure, configure.ac: Remove libnsl as a required lib for + libiksemel to work. This change was already made in the trunk. + (issue #8762) + + * /: Blocked revisions 52137 via svnmerge ........ r52137 | russell + | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a + seg fault when running this application with no arguments from + AGI. (issue #8905, junky) ........ + + * include/asterisk/dial.h: Fix the formatting of doxygen comments + to properly indicate that the comment documents the previous + entity, as opposed to the next one. + +2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com> + + * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 + line updated check_expr via 8322 (refactoring of expression + checking impl); elfring contributed a nice code reorg, I + contributed some time to get it working again, better messages + ........ + +2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com> + + * main/dial.c (added), apps/app_page.c, main/Makefile, + include/asterisk/dial.h (added): Merge in dialing API and the + app_page that uses it. (issue #BE-118) + + * channels/chan_sip.c: Fix changing channel formats when joint + capability changes and there are no audio formats... I didn't + break it originally! (issue #8535 reported by ivoc) + +2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com> + + * configure: rebuild configure script to reflect last chan_h323 + related changes. + +2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: added fix from #8899 + + * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 + Jan 2007) | 1 line fixed the busy problem (dialstatus was not + busy when we called a busy extension) ........ + +2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Show capabilities *and* preference in + general settings in "sip show settings" (reported by Clona/Telio + - Thanks!) + +2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz> + + * acinclude.m4: Allow x64 builds of H.323 (please, rebuild + configure) + +2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 51843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | + 6 lines Fix an issue related to synchronization of recordings + when using Monitor(). The bug is a miscalculation of the amount + to seek the stream for writing to disk when the number of samples + coming in and out of a channel do not match up. (issue #8298, + #8887, report and patch by guillecabeza, patch files created and + testing done by whoiswes) ........ + + * apps/app_while.c, /: Merged revisions 51828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | + 4 lines Don't set a new value for the END_ variable on the + channel before using the old value. If you do, it will lead to + accessing a memory address that has been free()'d. (issue #8895, + arkadia) ........ + +2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com> + + * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, + channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_alsa.c, + channels/chan_gtalk.c, channels/chan_iax2.c: Update channel + drivers to use module referencing so that unloading them while in + use will not result in crashes. (issue #8897 reported by junky) + +2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com> + + * main/manager.c: Fix some bugs in process_message(). The manager + session lock needs to be held when sending some sort of response, + or calling one of the manager action callbacks. This resolves an + issue where people using the GUI would get random crashes when + they start clicking around a lot. (issue #8711, reported and + debugged by zandbelt) + + * main/http.c: Fix setting the default port of 8088 on 64-bit or + big-endian machines. + + * main/manager.c: When traversing the list of manager actions, the + iterator needs to be initialized to the list head *after* locking + the list. Also, lock the actions list in one place it is being + accessed where it was not being done. + +2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com> + + * res/res_features.c: this mod from 8593 (dstchannel in cdr is + empty when transfer call). + + * main/callerid.c: via 8748 (callerid.c loses name when returning + PRIVATE_NUMBER flag), the user suggested this mod, saying it + would allow 'WITHHELD' to appear in the name field, which would + be useful + +2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | + 6 lines * more additions to make the RESTART message work * added + fix for misdn_call to allow SETUPs with empty extensions, + replaced the strtok_r functions with strsep for that (inspired by + Sandro Cappellazzo, thanks) ........ r50506 | crichter | + 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get + L2 UP, the L1 is UP definitely too, so we set the L1 state up as + well. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: manually merged r49922 and r50335, because + of conflicts. this commint includes addition of the ISDN RESTART + Message + +2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk + startup if h323 configuration file not found (reported by + mithraen) + +2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Only change audio formats on the channel if + we have an audio format to change to. (issue #8535 reported by + ivoc) + + * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan + 2007) | 2 lines Yield before reading from zaptel timing source + under Solaris so that other threads get a chance to do things. + (issue #7875 reported by bob) ........ + +2007-01-22 19:41 +0000 [r51411] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 51410 via svnmerge ........ r51410 | russell + | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge + codec_zap support for the transcoder card. This is a standalone + codec module so it will not affect anything else. ........ + +2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: This fixes 8836, according to dnatural + +2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com> + + * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan + 2007) | 2 lines Move filestream creation to Mixmonitor loop. This + will prevent a blank file from being created if no frames ever + pass through to be recorded. (issue #7589 reported by + steve_mcneil) ........ + + * /: Blocked revisions 51359 via svnmerge ........ r51359 | file | + 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly + declare what codecs are supported by default globally since using + a bitmask for all may include ones we don't need. (issue #8357 + reported by gknispel_proformatique) ........ + +2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com> + + * configs/say.conf.sample: Fix Italian numeral support in say.conf + for "_[2-9]00" case. "2131" would've translated to something + along the lines of (pardon my..Italian {or lack thereof}) + "duecentocentotrentuno", which makes no sense at all. + + * configs/say.conf.sample: Fix German language support in say.conf + Properly support 21, 31, 41, 51, 61, 71, 81, and 91. + einundzwanzig has the same format as zweiundzwanzig (as do all + other "_ZX" spoken numerals) Fix support for numbers in the + 10,000,000 to 99,999,999 range. Add support for numbers in the + 100,000,000 to 999,999,999 range. + +2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Remove an unused instance of an unnamed enum. + + * apps/app_meetme.c: Remove another duplicated definition + + * apps/app_meetme.c: Remove a variable that was declared twice. + + * codecs/gsm/Makefile: Add a couple more processors that need + optimizations excluded. (issue #8637) + + * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. + AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same + thing. So, a digit would have been interpreted incorrectly here. + Since the channel driver will always have the begin and end + callbacks called for a digit, only support the button-down and + button-up messages. + + * .cleancount: Bump the cleancount since my last commit changed the + channel structure. + + * channels/chan_oss.c, main/rtp.c, main/channel.c, + channels/chan_phone.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_features.c, + channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, + channels/chan_zap.c, channels/chan_local.c, main/frame.c, + channels/chan_sip.c, channels/chan_agent.c, + include/asterisk/channel.h, channels/chan_gtalk.c, + channels/chan_iax2.c: Merge the changes from the + /team/group/vldtmf_fixup branch. The main bug being addressed + here is a problem introduced when two SIP channels using SIP INFO + dtmf have their media directly bridged. So, when a DTMF END frame + comes into Asterisk from an incoming INFO message, Asterisk would + try to emulate a digit of some length by first sending a DTMF + BEGIN frame and sending a DTMF END later timed off of incoming + audio. However, since there was no audio coming in, the DTMF_END + was never generated. This caused DTMF based features to no longer + work. To fix this, the core now knows when a channel doesn't care + about DTMF BEGIN frames (such as a SIP channel sending INFO + dtmf). If this is the case, then Asterisk will not emulate a + digit of some length, and will instead just pass through the + single DTMF END event. Channel drivers also now get passed the + length of the digit to their digit_end callback. This improves + SIP INFO support even further by enabling us to put the real + digit duration in the INFO message instead of a hard coded 250ms. + Also, for an incoming INFO message, the duration is read from the + frame and passed into the core instead of just getting ignored. + (issue #8597, maybe others...) + + * main/asterisk.c: Merged revisions 51300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | + 4 lines Fix a memory leak on command line tab completion. The + container for the matches was freed, but the individual matches + themselves were not. (issue #8851, arkadia) ........ + +2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com> + + * channels/chan_zap.c: chan_zap compiles without libpri after + committing 7877 patch + + * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) + | 3 lines issue 7877: chan_zap module reload does not use + default/initialized values on subsequent loads. Reset + configuration variables to default values prior to parsing + configuration file. ........ + +2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com> + + * /: block this patch since it is already here + +2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c, main/channel.c, main/pbx.c, + funcs/func_strings.c, main/app.c: Add some more checks for + option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, + patch(es) by tgrman + +2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com> + + * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: + Ensure that the locations given to the Asterisk configure script + for ncurses, curses, termcap, or tinfo are further passed along + to the editline configure script. This fixes some + cross-compilation environments. (issue #8637, reported by ovi, + patch by me) + +2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 + Jan 2007) | 2 lines If a timezone is not specified, assume + localtime (instead of gmtime) (Issue #7748) ........ + +2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com> + + * apps/app_speech_utils.c: Only start timeout once we reach the end + of the files to play back. + +2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com> + + * main/cli.c: Fix an issue with file name completion in "module + load" and "load". Issue 8846 + +2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Copy MOH settings when calling a peer so + that if they put someone on hold or get put on hold themselves + they get the right music class. (issue #8840 reported by mdu113) + +2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com> + + * main/channel.c: Fix an issue with deprecated commands + +2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 + Jan 2007) | 2 lines Document all the fields, including the + indication that "uniqueid" should not be renamed. ........ + +2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com> + + * main/manager.c: Make the "hasmanager" option in users.conf + actually have an effect. (issue #8740, LnxPrgr3) + +2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Build the IMAP remote directory string + better and properly. Fix an issue with encoding the GSM voicemail + when attaching to the voicemail. (issue #8808 reported by + akohlsmith) + + * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. + (issue #8840 reported by mdu113) + +2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com> + + * funcs/func_odbc.c: Fix some instances where when loading + func_odbc, a double-free could occur. Also, remove an unneeded + error message. If the failure condition is actually a memory + allocation failure, a log message will already be generated + automatically. + + * channels/chan_zap.c: Instead of dividing the offset by 2 + directly, make it more clear that the offset is being scaled by + the size of the elements in the buffer. (Inspired by a discussing + on the asterisk-dev list about this code) + + * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | + 3 lines Move the check for a failure of ast_channel_alloc() to + before locking the pvt structure again. Otherwise, on a failure, + this will cause a deadlock. ........ + +2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, main/utils.c: Merged revisions 51194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) + | 4 lines When ast_strip_quoted was called with a zero-length + string, it would treat a NULL as if it were the quoting character + (and would thus return the string in memory immediately following + the passed-in string). ........ + +2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: re-add "password" for realtime voicemail + +2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Return the correct result when directly writing out a + packet so that the core doesn't then decide to handle it the + regular way again. (issue #8833 reported by rcourtna) + +2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_voicemail.c: a few more coding style cleanups and one + bug fix (from AnthonyL) + +2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the + scheduler callback. + +2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com> + + * main/rtp.c: Fix issue with dtmf continuation packets when the + dtmf digit is 0... Issue 8831 + + * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with + IMAP storage and realtime voicemail. Also update the vmdb sql + script for IMAP specific options. Issue 8819, initial patches by + bsmithurst (slightly modified by me) + + * doc/voicemail_odbc_postgresql.txt: change documentation to + reflect new procedure in 1.4/trunk + +2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions + 51161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) + | 2 lines Add documentation walkthrough on getting Postgres to + work with voicemail (from Issue 8513) ........ + + * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) + | 2 lines Postgres driver doesn't like a NULL pointer when + retrieving the length (Bug 8513) ........ + +2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c: minor things i missed before i get jumped + on + +2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com> + + * /, res/res_features.c: Merged revisions 51145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 + lines Return previous behavior. ParkedCalls will be able to do + DTMF based transfers again. trunk however will get an option to + allow this to be set on/off. (issue #8804 reported by nortex) + ........ + +2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com> + + * main/file.c: Display more useful output when streaming files. + Include the channel name to which the file is being played. Issue + 8828, patch by junky. + +2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 + lines Add none as a valid callgroup/pickupgroup option. I + consider it a bug that it would inherit it all the way down and + not have any way to reset it to nothing - so that's why it is in + 1.2. (issue #8296 reported by gkloepfer) ........ + +2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com> + + * main/config.c: It is possible for the config pointer to be NULL + here, so it needs to be checked before dereferencing it. + +2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for + changing voicemail password in users.conf from voicemail main, + written by AnthonyL bug #8436 + +2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com> + + * Makefile.rules: Filter out a few CFLAGS that are not valid + CXXFLAGS. + +2007-01-15 23:10 +0000 [r50988] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /: Blocked revisions 50987 via svnmerge ........ r50987 | + tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines + Check return value before dereferencing (Bug 8822) ........ + +2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 + | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 + lines Solves issue with forwarding voicemails from folders other + than inbox. patch by anthonyl. ........ + +2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com> + + * main/asterisk.c: re-add deprecated "show version" CLI command. + +2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com> + + * main/manager.c: Move event processing into do_message so that it + gets executed again when events are tripped. + +2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com> + + * configure, include/asterisk/autoconfig.h.in, main/Makefile, + configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the + ACX_PTHREAD macro from the Autoconf macro archive for setting up + compiler pthreads support... should improve portability to + platforms with unusual pthreads requirements + +2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com> + + * main/astmm.c: Add missing newlines for two memory CLI commands. + +2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, + main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, + main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, + main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, + main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, + main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, + main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, + main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, + main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, + main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, + main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, + main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, + main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, + main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, + main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, + main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 + Jan 2007) | 2 lines Bug 8814 - db should look for its header + using a relative path, instead of the system path (Fixes FreeWRT) + ........ + +2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, build_tools/make_sample_voicemail (added): when + building the sample greetings for maibox 1234@default during + 'make samples', build a greeting for each language and file + format the user selected to install with menuselect (reported by + Brian Capouch on asterisk-dev) + +2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Only write a frame out to the channel if one + exists. There are cases where one may not and would therefore + cause the channel driver to segfault. (issue #8434 reported by + slimey) + + * res/res_snmp.c: Only join the snmp thread on an unload if the + thread is actually running. (issue #8810 reported by junky) + +2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com> + + * configs/voicemail.conf.sample: Update documentation to state that + you shouldn't use realtime static with voicemail.conf + +2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com> + + * main/manager.c: We need to check for res being 0 in do_message + itself, otherwise our headers will get lost. + +2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com> + + * main/pbx.c, /: Merged revisions 50561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) + | 2 lines minor documentation clarification ........ + +2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Remove check for channel state as it can + definitely be something other then ring, and also clean up the + code a bit. This should solve the parking issues and maybe some + attended transfer issues people have been seeing. + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add + support to see whether NAT was detected (yay symmetric RTP) and + also add a check in chan_sip so that if NAT has been detected and + the reinvite behind nat option has been turned off, then just do + partial bridge. (issue #8655 reported by mnicholson) + + * apps/app_speech_utils.c: Merge speech-multi branch which adds + support for joining multiple sound files together to be played + one after another in SpeechBackground. + + * main/config.c: Fix parsing when using something like ldap + settings. (done by anthonyl) + + * channels/chan_sip.c: Fix chan_sip not working issue. Let's not + prematurely return 0. (issue #8783 reported by st41ker) + +2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com> + + * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made + it fail to load if the config file existed. Issue 8777 + +2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 + lines Add another return value to dial_exec_full that indicates + execution is going to continuing at a new + extension/context/priority and to just let it slide. (issue #8598 + reported by jon) ........ + + * main/pbx.c: Ensure data's existence before trying to access it. + (issue #8774 reported by rcourtna) + +2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 50227 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | + 6 lines Make the number that represents the major version number + a single digit instead of 2. Using two digits makes it an octal + number when put into version.h, which breaks the compilation of + any out of tree module that checks the version for any version + after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev + mailing list, who gave credit to vihai for pointing it out) + ........ + +2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com> + + * main/cli.c: Re-add CLI command that should have only been + deprecated in 1.4. Thanks kshumard! (reported in person, so no + associated issue #) + +2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) + | 4 lines The advent of realtime has enabled people to use commas + in the fullname field. This could cause an issue with sending + voicemails, when the field is unquoted. (Issue 8595) ........ + +2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - handle re-invites properly in sip_hangup() + - Add some invitestate status changes just to be sure + +2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, + where it wouldn't ever parse a password, since it was using + "secret" instead of "password" Issue 8761, reported by and patch + suggestion from ssokol. + +2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com> + + * apps/app_senddtmf.c: we can't unlock a channel if we cant find + it. - AnthonyL bug #8741 + +2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Disable the more intense packet2packet bridging until + the bugs can be worked out. + +2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 + re-invite This is not a fix, but adding an error message to tell + the admin that we have a bad configuration. We should not send + T.38 re-invites to devices that can't handle it (with the current + architecture where you have to hard-code t.38 support per + device). To really fix this, we need to figure out a way to tell + the incoming call that the re-invite failed, so we can signal + failure on that end and go back to the original call. + + * channels/chan_sip.c: Issue #8524, support multiple via header + values (tardieu) Thanks! + + * channels/chan_sip.c: We only need one forward declaration + + * channels/chan_sip.c: Issue 8735: Terminate state when extension + is unavailable for subscription + +2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 + lines Ensure we use the default refresh value of 60 if the remote + server does not send one. (issue #8746 reported by maethor) + ........ + +2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com> + + * configure, configure.ac: since we use AC_PATH_TOOL to find tools, + we should use the results it provides for us (reported by Brian + Capouch on the asterisk-dev list) + +2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) + | 2 lines If openstream fails, then we crash (Issue 8564) + ........ + + * channels/chan_sip.c: Second condition was a subset of the first, + so hold was never decremented, thus hint stayed stuck (Issue + 8747) + +2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com> + + * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping + byte of allocated memory! This looks like it may have been a + chicken/egg scenario.. You had to call a cleanup func, because + everything was allocated. Then since you had to call a cleanup + func, you were forced to allocate - ie; strdup(""). + +2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com> + + * configure, acinclude.m4: one more time... + + * configure, acinclude.m4: proper fix for r49712 + + * configure, acinclude.m4: if --with-foo=<path> is specific for a + configure option, ensure that it is used for header file checking + as well + + * main/manager.c: ast_func_read() needs a writable copy of the + function name to be passed + +2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and + chan_zap also depend on zaptel. This fixes an issue (8727) with + zaptel being in a different directory, using --with-zaptel. + +2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com> + + * main/manager.c: don't 'consume' the params list before we try to + use it again + + * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, + main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, + main/db.c, channels/chan_zap.c, channels/chan_sip.c, + apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, + utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, + apps/app_queue.c, res/res_jabber.c: reduce stack consumption for + AMI and AMI/HTTP requests by nearly 20K in most cases + +2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Don't keep repeating the warning over and over + when the end of the call is reached. (issue #8724 reported by + xrg) + +2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_iax2.c: Merged revisions 49635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) + | 2 lines ensure that threads which are supposed to be detached + (because we aren't going to wait on them) are created properly + ........ + + * channels/chan_iax2.c: revert the dynamic_list insertion change... + that was not the right thing to do + + * channels/chan_iax2.c: create the IAX2 processing threads as + background threads so they will use smaller stacks when we create + a dynamic thread, put it on the dynamic_list right away so we + don't lose track of it + +2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: It's possible for the iax2 pvt to + disappear, so if it has... don't bother looking for dpentries. + +2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/threadstorage.h, main/asterisk.c, + build_tools/cflags.xml, include/asterisk.h, main/Makefile, + main/threadstorage.c (added), main/utils.c: add support for + tracking thread-local-storage objects that exist via + 'threadstorage' CLI commands + +2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com> + + * main/config.c: Only free comments and line buffer once we reach + the first level. (issue #8678 reported by ssokol, fixed by + anthonyl) + +2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com> + + * channels/iax2-parser.c, main/frame.c: don't mark these + allocations as 'cache' allocations when caching has been disabled + + * channels/iax2-parser.c: if we're going to decrement the frame + count when we free a frame, we should inrement it when we create + one :-) + + * channels/iax2-parser.c, channels/iax2-parser.h, + channels/chan_iax2.c: only do IAX2 frame caching for voice and + video frames + + * main/frame.c: don't do frame header caching in the core if + LOW_MEMORY is defined + + * channels/iax2-parser.c: don't define this type either if + LOW_MEMORY is enabled + +2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 + | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 + lines converted a lot of 256 to PATH_MAX and some white space + fixes. ........ + +2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com> + + * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode + + * codecs/Makefile: make building of codec_gsm against the system + GSM library actually work + +2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 + | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 + lines good catch russell sorry i missed that. fix magic number + with proper sizeof ........ + +2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com> + + * funcs/func_realtime.c: Fix the REALTIME() dialplan function. + ast_build_string() advances the string pointer to the position to + begin the next write into the buffer. So, this pointer can not be + used to copy the contents of the string later. The beginning of + the buffer must be saved. Interestingly enough, this code could + not have ever worked. (Pointed out by Sebb on IRC, thanks!) + +2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 + | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 + lines When using ODBC_STORAGE VoicemailMain doesn't create the + subdirectories for a mailbox such as the INBOX directory. this + patch solves that problem, was written by anthony be-125 ........ + +2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, + /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, + configs/misdn.conf.sample: Merged revisions + 48319,48321,48467,48552,48576,49135,49303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | + 1 line changed a few debugs to higher debug levels ........ + r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | + 1 line added the export and import of the MISDN_ADDRESS_COMPLETE + Variable to inidcate wether the extension is already completely + dialed or if there might come additional digits by information + elements. also added some docs for that. ........ r48467 | + crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line + removed FIXUP state. added check for channel allocation conflict + when we create a setup while the other site creates a setup on + the same channel, besides the check we resolve this conflict. + ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 + Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a + preselected channel we just accept it, even when we're NT. added + some checks for segfaults. ........ r48576 | crichter | + 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we + reject a channel, because it's in use already, we shouldn't + process the setup anymore. made the channel allocation a bit + easier and more understandable, removed a few unused lines + ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 + Jan 2007) | 1 line added check for channel ranges in the + set/empty channel functions. set pmp_l1_check default to no. + added misdn restart pid cli command. added cleaning of channel + when we send a RELEASE_COMPLETE. ........ r49303 | crichter | + 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added + check for bridging in misdn_call to avoid setting + echocancellation when 2 mISDN channels are involved and when + bridging is set. That lead to a kernel panic before under + different situations, because we switched about 2 times between + hardware bridging and echocancelation * readded MISDN_URATE + variable which got lost before, this should make app_v110 work + again * fixed typo ........ + +2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, Makefile.rules: various Makefile improvements to get + chan_vpb (and any other C++ modules) to build properly + +2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Check pvt structure presence before passing + to send_command. This gets rid of the irritating message about a + packet without pvt structure. This happens because the scheduled + item is getting cancelled at almost the exact moment it is + getting executed. + +2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com> + + * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, + pbx/ael/ael.flex: This is a slight modification to Josh's edits + for #8579; both files edited were the produced by flex; so the + source files need to be changed instead, and the generated files + regenerated. + +2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always + enabled at that point in the code + +2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com> + + * main/pbx.c: Allow fractions of a second in the Wait() + application, like it says it allows. + +2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: remove comment that is unrelated to this + function + +2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net> + + * configs/features.conf.sample: Adding note on effect of + applicationmap features on re-invites + +2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, + configure.ac, codecs/codec_zap.c: check specifically for VLDTMF + and transcoding support in the system's Zaptel installation, and + make only the modules that need those features dependent on them + (this will allow building the other Zaptel-using parts of + Asterisk against older versions of Zaptel or those on other + platforms that haven't caught up yet to the Linux version) + + * Makefile: use a simpler (and portable) method to ensure that + menuselect is built as a host binary + + * Makefile: revert this change until a better solution can be + found... 'env -i' was not being used properly, but even when + changed to do so, this process fails during cross-compilation + because the menuselect build still sees 'CC' as set to the + cross-compiler + +2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: remove incomplete implementation of dnsmgr. + Let's fix this in trunk. + +2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com> + + * pbx/pbx_config.c: IAX has been deprecated for quite some time so + we had better use IAX2 when creating the dial string for users. + (issue #8697 reported by ssokol) + + * channels/chan_zap.c: Use asprintf to build the channel names + instead of custom function. I believe the custom function is + doing some things that are not portable across all + implementations. (issue #8570 reported by hterag & issue #8692 + reported by nicolasg) + + * main/rtp.c: If the Packet2Packet bridge is being broken because + of a masquerade then attempt to read a frame in so the masquerade + actually happens. Otherwise weirdness will occur. (issue #8696 + reported by kjotte) + + * channels/chan_iax2.c: Initialize the packet queue in load_module + instead of just declaring the list with the default value. (issue + #8695 reported by ssokol) + +2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have + comma args converted to vertical bars. I hope this change does + little harm. + +2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com> + + * /: put this value into the correct property + + * /, BUGS: Merged revisions 49045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) + | 2 lines location of the bug posting guidelines has changed + ........ + + * sample.call: simple commit to test CIA integration + +2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com> + + * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me + + * main/http.c: saw this in passing... fix a small typo + +2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile: new versions of sounds + +2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com> + + * main/http.c: make the uris_lock a rwlock instead of a mutex lock + - needs to be forward ported to trunk + +2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Backport support for read/write locks. + +2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com> + + * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, + pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, + pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, + pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk + from the ael stuff. Also, threw in a minor fix to frame.c to + avoid build-killing compiler warnings. + +2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com> + + * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not + available when LOW_MEMORY is used and things are being built in + the utils directory, so we need to resort to the old method of + strncpy. (issue #8579 reported by mottano) + +2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com> + + * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, + main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, + main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, + main/http.c, main/logger.c: since these variables all have static + duration, none of them need initializers (they default to zero + anyway) + + * include/asterisk/options.h, main/asterisk.c, main/file.c: move + extern declaration for this option to a header file where it + belongs provide an initial value for 'languageprefix' option, + instead of relying on randomness to provide a useful value + +2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Only include acl.h and lock.h once + + * channels/chan_sip.c: Only set rfc2833compensate flag once + (handle_common_options) + + * channels/chan_sip.c: - Remove checking for T38 options twice. + Keeping them in handle_common_options + +2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: make the option actually match the + documentation + + * channels/iax2-parser.c, include/asterisk/utils.h, + include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show + memory' and 'show memory summary' to distinguish memory + allocations that were done for caching purposes, so they don't + look like memory leaks + +2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more + politically correct + + * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy + cisco MWI support. Normally we try not to change our software for + bugs in other devices. But in this case, the Cisco phones are so + widespread so we try to implement a fix while waiting for a + bugfix from Cisco. + + * channels/chan_sip.c: - Make sure handle_common_options return 1 + when we found a common option - Move uncommon (only global) + option away from handle_common_options Reported by rizzo. Thanks! + + * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before + re-sending invite with auth. + + * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. + (rizzo, #8600) + +2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com> + + * apps/app_meetme.c: Get rid of a needless memory allocation and + only create a conference structure in find_conf_realtime if data + was read from realtime. (issue #8669 reported by robl) + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an + API call that initializes an RTP structure. We need this because + chan_sip is cheeky and uses a temporary RTP structure for codec + purposes, and the API calls that are used rely on the lock. + (Pointed out on asterisk-dev by Andy Wang) + + * configure, configure.ac: Clean up autoconf file (gets rid of + warnings seen when rebuilding configure) and rebuild configure. + +2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com> + + * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | + 6 lines Fix an error introduced by copying and pasting the + handling of the >= operator for the MATH function. If a single + equal sign was used as an operator, the function would treat it + is as if it were the >= operator. Now, it properly handles it as + an invalid operator. (issue #8665, patch by tempest1) ........ + + * channels/chan_oss.c: Fix a typo in an error message that + indicated that the MGCP channel type could not be registered, + instead of the correct type, OSS. + + * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | + 3 lines Check for the proper return value on an error in a call + to mmap(). This was reported by Andy Wang on the asterisk-dev + list. Thanks! ........ + + * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | + 3 lines Remove a couple of misplaced dots in log messages. This + was reported by Andrea Spadaccini on the asterisk-dev mailing + list. ........ + + * main/http.c: Implement locking for the list of URI handlers to + make it thread-safe. + +2006-12-23 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.4.0 released. + +2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com> + + * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. + + * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia + +2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com> + + * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 + lines Add new silence sound files to the spec for Redhat. (issue + #8652 reported by alvaro_palma_aste) ........ + +2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage + builds. + + * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so + it is then passed to the IMAP store file function. (issue #8614 + reported by punknow) + + * doc/snmp.txt: find is not the same as bind when it comes to + documentation. (issue #8626 reported by johann8384) + +2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com> + + * channels/Makefile: suppress compiler warnings in this module + until it can be improved + +2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 + lines Free localuser structure when we fail to dial (issue #8612 + reported by rizzo) ........ + +2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org> + + * apps/app_sms.c: fix a bogus datalen in the frames generated by + app_sms (causing noisy output if you listen to the output!) This + affects trunk as well, whereas 1.2 is ok. + +2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com> + + * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable + type for these unixODBC API calls, eliminating warnings on 64-bit + platforms that use the 'new' 64-bit types for ODBC API calls + +2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com> + + * Makefile: Use env -i to start a fresh environment when going to + build menuselect. This is more portable then using unset. (issue + #8543 reported by jtodd) + +2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/channel.h: unbreak the macro used for + incrementing the frame counters. I don't know when the bug was + introduced, but with the typical usage c->fin = + FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects + trunk as well (fix coming). + +2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Put thread into proper list if we abort + handling due to an error, and also hold the lock while putting it + back into the proper idle list so we don't prematurely get a + signal. (issue #8604 reported by arkadia) + +2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com> + + * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, + utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, + utils/ael_main.c: remove some now-unnecessary explicit includes + of autoconfig.h clean up per-file dependencies during 'make + clean' + + * build_tools/prep_tarball: need an additional argument here to + make the downloads actually occur + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep + these calls from thinking they have multiple arguments + + * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, + funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, + main, codecs/gsm, pbx, res, channels, codecs, utils, agi, + main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: + simplify dependency tracking system, using the compiler's + built-in method for generating them, and only doing dependency + tracking if developer mode is enabled via the configure script + + * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we + really, really have to have autoconfig.h included before all + other headers (especially system headers), the Makefile will now + force it to happen (this will fix build problems with files like + ast_expr2f.c, where we can't control the inclusion order in the + file itself) + + * funcs/func_curl.c: instead of initializing the curl library every + time the CURL() function is invoked, do it only once per thread + (this allows multiple calls to CURL() in the dialplan for a + channel to run much more quickly, and also to re-use connections + to the server) (thanks to JerJer for frequently complaining about + this performance problem) + +2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Turn payload_lock into bridge_lock and make it + encompass all RTP structure contents that may relate to bridge + information, including who we are bridged to. + + * channels/chan_iax2.c: Hold call structure lock in places where a + qualify or peer action can destroy it. + + * channels/chan_iax2.c: Lock network retransmission queue in all + places that it is used. + +2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported + from 1.2) + + * channels/chan_sip.c: Update to latest IANA spec + +2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Use a wakeup variable so that we don't wait + on IO indefinitely if packets need to be retransmitted. + + * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP + structure can change AFTER a bridge has started. This comes from + the packet handling of the SIP response when indication that it + was answered has been sent. Therefore we need to protect this + data with a lock when we read/write. (issue #8232 reported by + tgrman) + + * main/rtp.c: Remove direct RTCP bridging. I've come to the + conclusion that we should handle this through the core and not + just forward it on. Should solve a few bugs. + +2006-12-12 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.4.0-beta4 released. + +2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This + is the way it should have been done. + +2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com> + + * sounds/Makefile: new sounds package with 100% more silence + + * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge + from https://svn.digium.com/svn/asterisk/branches/1.2 ........ + r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) + | 4 lines app_externalivr needs a real silence file, and + additional changes to add silence files into core instead of + extra patch provided by bug 8177 with minor additions. ........ + +2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Return non-existant callerid handling to + that which it was before. In 1.4 and trunk callerid can be + allocated but not have any contents so we have to use + ast_strlen_zero before passing it to the relevant functions. + (issue #8567 reported by pabelanger) + +2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * funcs/func_strings.c: STRFTIME() does not actually require an + argument (issue 8540) + +2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Merge in my latest RTP changes. Break out RTP and + RTCP callback functions so they no longer share a common one. + + * apps/app_meetme.c: Use the correct API call to say a device state + changed. (Yes, I'm a nub.) + + * apps/app_meetme.c: Don't access the conference structure after it + has been freed. + +2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, + res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, + apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) + | 5 lines When doing a fork() and exec(), two problems existed + (Issue 8086): 1) Ignored signals stayed ignored after the exec(). + 2) Signals could possibly fire between the fork() and exec(), + causing Asterisk signal handlers within the child to execute, + which caused nasty race conditions. ........ + +2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com> + + * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 + line This version applies the patch suggested by stevens in bug + 7836 (make inbound channel RINGING state consistent with other + channels). ........ + +2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Use locking when accessing the + registrations list. This list is not actually used very often, so + the likelihood of there being a problem is pretty small, but + still possible. For example, if the CLI command to list the + registrations was called at the same time that a reload was + occurring and the registrations list was getting destroyed and + rebuilt, a crash could occur. In passing, go ahead and convert + this list to use the linked list macros. + + * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell + | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use + locking when accessing the registrations list. This list is not + actually used very often, so the likelihood of there being a + problem is pretty small, but still possible. For example, if the + CLI command to list the registrations was called at the same time + that a reload was occurring and the registrations list was + getting destroyed and rebuilt, a crash could occur. ........ + +2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 + Dec 2006) | 3 lines Ensure that the file position is not + incremented beyond the total number of files available for + playback. (issue #8539, ulogic) ........ + +2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com> + + * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that + killed bug 8423 -- OriginateSuccess and OriginateError incomplete + channel name. May it rest in peace. + +2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being + retransmitted to Asterisk + +2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com> + + * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 + Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option + in the sample configuration file. (issue #8526, arkadia) ........ + +2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Don't send Contact on MESSAGE + +2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com> + + * configure.ac: Fix curl version number testing to be much more + friendly to non-bash shells. Issue 8508, patch by me. This + *SHOULD* be POSIX compliant now.. + +2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Merging the invitestate-1.4 branch after + successful testing. Will check if I can solve this with less + changes in 1.2. + + * configs/sip.conf.sample: Add missing s from another repository. + (thanks jcmoore!) + + * configs/sip.conf.sample: Updating sip.conf.sample with + information about T38 not working when chan_local or chan_agent + is involved in the call. I don't know how big a fix that would be + to solve, but this is the current state of affairs. (Chan_sip + currently checks if the other side of the bridge has a SIP tech. + We could/should implement another check, possibly for udptl_write + or some flag in the ast_channel structure). + +2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Oops, forgot to release the odbc handle + + * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) + | 6 lines If the recording in the database is too large, it will + fail to retrieve with an mmap error. Not too sure why this + doesn't happen when we put it in the database, also, but since + that doesn't seem to be broken, I'm not going to fix it (at least + until someone reports it). Solution is to ask for the file in + smaller chunks. (Bug 8385) ........ + +2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Fix an issue which didn't allow + unavail/greet/busy/etc messages from being saved into ODBC (and + probably IMAP). + + * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell | + 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert + change from 8016 - this breaks other stuff... Needs further + review. Tip: When you've reported a bug about something and + somebody has put up a patch for it.. It's not a good idea to open + a completely new bug and say that something is broken because of + the patch in the other bug - PLEASE mention something in the bug + where the patch was actually created. ........ + + * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell | + 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an + issue where a message isn't saved correctly when using ODBC + storage and reviewing a message. Issue 8016 - patch by sokhapkin. + ........ + +2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 48233 via svnmerge ........ r48233 | file | + 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the + generic bridge tells us not to retry, and we have a frame to spit + out then break the bridge. Props to markit in #asterisk-bugs for + bringing this up. ........ + +2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com> + + * configs/voicemail.conf.sample: Add documentation to + voicemail.conf.sample for ODBC storage. Issue 8499 - patch by + blitzrage. + + * doc/snmp.txt: Attempt to document some of the dependencies that + are needed for net-snmp Issue 8499 - initial patch by blitzrage. + +2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com> + + * sounds/Makefile: When "fetch" is in use, instead of "wget", + --continue is not a valid option. (issue #8451) + +2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - Removing one of two pieces of code to + handle 481 response on INVITE - Move handling of REFER response + to handle_response_refer() + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, + configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax + transmission happens - Encapsulate RTP timers in the rtp + structure so we have one for video and one for audio The video + one is not used in 1.4, really. Will be used for RTP keepalives + when we can send something that video phones support in the RTP + stream. I now this is a big architectual change at this stage for + 1.4, but decided it was needed to avoid future bug reports. - + Document the RTP NAT keepalive option in sip.conf.sample Issue + 7679 in the bug tracker. Please test. + +2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com> + + * include/asterisk/utils.h: Backport the comment containing the + warning regarding the limitations on the usage of this function. + It is thread safe, but not technically reentrant. + +2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) + | 2 lines if Dial() is going to send music-on-hold to the calling + party, it has to send PROGRESS first to ensure that the reverse + audio path has been setup first (BE-106) ........ + +2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com> + + * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: + FreeBSD 6.1 does not include wget by default. However, it has + fetch which will work just fine for our purposes of downloading + the sounds packages. So, check for both wget and fetch and the + configure script and use what was found to download them. If + neither one was found, and sound packages are selected that must + be downloaded, the install process will print out an informative + error message indicating the situation. Also, fix a couple places + where "make" was hard coded into some output messages by + replacing them with the $(MAKE) variable. (issue #8451, initial + patch by pabelanger, with additional modifications by me) + +2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 48183 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 + lines Fix a small typo - issue 8848, reported by pabelanger + ........ + +2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/cli.c: Double-unlock error (reported by blitzrage on IRC) + +2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the + "limitonpeers" patch from trunk, to fix a lot of issues with + queues and SIP device states - Remove support for T.38 early + media, since it's impossible. (Two patches in one - extra friday + evening offer due to being off line from svn today... :-) + +2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not + do a partial bridge for Google Talk since we need to handle STUN. + (issue #8448 reported by phsultan) + +2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Issue 8319 - change noncecount before + using it. + +2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 48161 via svnmerge ........ r48161 | file | + 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't + write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel + driver. (issue #8390 reported by hselasky) ........ + + * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 + lines Only print out debug message if bridged channel is not + NULL. (issue #8412 reported by jubilex) ........ + + * /, res/res_features.c: Merged revisions 48154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 + lines Do not listen for DTMF on the bridge that comes into + existence when ParkedCall is executed. This means native bridging + can now occur for this. (issue #8406 reported by kebl0155) + ........ + + * main/cdr.c, /: Merged revisions 48151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 + lines Print certain CDR messages out at the NOTICE level versus + WARNING since they can occur when used with the CDR applications + and are perfectly fine. (issue #8367 reported by dartvader) + ........ + + * /: Blocked revisions 48146 via svnmerge ........ r48146 | file | + 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember + the pointer to the allocated block of memory so that we can free + it and not cause a memory leak. (issue #8449 reported by arkadia) + ........ + + * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov + 2006) | 2 lines Document 'port' for SIP peers, came up because of + the current mailing list thread. (issue #8450 reported by + blitzrage) ........ + +2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net> + + * doc/manager.txt: Explain status reports and make codefreeze more + happy :-) + + * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by + GS 487 adapter without CSEQ on separate line in the REGISTER + request. Imported from 1.2. + +2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in + mm_login. (issue #8420 reported by slimey) + +2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Explain the use device status system + implemented in SIP for subscriptions, queues and manager a bit + better. Like in 1.2, you will get more detailed information if + you set a call limit for a device. When the call limit is + reached, the status system will report a device as busy. For + queues, setting a call limit per SIP device is propably a + requirement. In most cases, it will work much better if you only + use type=peer and not type=friend. We might decide to backport + the new setting from trunk to apply all call limits to the peer + part of a friend only. + +2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, /: Merged revisions 48106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 + lines If the frame was duplicated before writing out then we need + to free it. (issue #8429 reported by edguy3) ........ + +2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. + +2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Don't crash if the mailstream was not + created. + +2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com> + + * Makefile: Export several more variables in top level Makefile. + Inspired by issue 8438. + +2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com> + + * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov + 2006) | 2 lines According to the research I have done we never + needed to include compiler.h in the first place so let's not! + (issue #8430 reported by edguy3) ........ + + * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 + lines Use the proper function to get the new message count + instead of always using the filesystem. (issue #8421 reported by + slimey) ........ + +2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 + Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) + ........ + +2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com> + + * main/manager.c: Remove a couple of unused variables (issue #8380, + casper) + +2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 + lines Do not reference the freed outgoing structure in the debug + message. (issue #8425 reported by arkadia) ........ + +2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Change logging message + +2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com> + + * funcs/func_cdr.c: might as well also document the raw values of + the flag vars + + * /, funcs/func_cdr.c: A little bit of func_cdr documentation + upgrade-- no bug# involved, although 8221 may have inspired it. + +2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 + and future releases, you can disable subscription support totally + or per peer in sip.conf with allowsubscribe = yes | no + +2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com> + + * main/translate.c: bug 8189 posted this fix for main/translate.c + for PLC + +2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 + Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. + beatufied some logs, changed some loglevels. changed the default + value of block_on_alarm ........ + +2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Don't allocate unused variable. + +2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Video will never reach Packet2Packet bridging and can + do more harm then good. + +2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: If we have the non standard G726-32 setting turned on + we want to return G726-32 to the SDP, not our AAL2 string. (issue + #8330 reported by voipgate) + +2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 + provisional response. Let's not treat that as early media. + (discovered at the AVTF meeting in Paris). + +2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Oops, merge missed release of odbc object + + * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) + | 2 lines Failing to trap -1 error from mmap causes segfault + (Issue 8385) ........ + +2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com> + + * main/frame.c, /: Merged revisions 47859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 + lines Don't forget to byte swap if we are exiting the smoother + feed early. (issue #8287 reported by arturs) ........ + + * /: Blocked revisions 47855 via svnmerge ........ r47855 | file | + 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free + history items at the end of use of the temporary SIP pvt + structure. (issue #8383 reported by benh) ........ + + * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists. + + * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c, + include/asterisk/channel.h: Use a separate variable in the + channel structure to store the context that the channel was + dialed from. (issue #8382 reported by jiddings) + +2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Explain properly how videosupport works. + Committ from Asterisk Video Task Force meeting in Paris! + + * /, channels/chan_sip.c: Make sure we destroy scheduled items and + not use them ever again after destruction (rizzo) + +2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header + contains angle brackets (the bug was only in a corner case where + the < was right after the opening quote, and the fix is trivial). + +2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com> + + * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially + pointed out by mrobinson. + + * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell | + 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a + couple of typos in applications.. Initially spotted by mrobinson. + ........ + +2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com> + + * /, doc/billing.txt: update documentation regarding IAX2 transfers + and CDRs Merged revisions 47776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) + | 2 lines update clearly wrong documentation regarding cdr_custom + ........ + +2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Compare technology using the pointers + instead of a straight comparison based on name. (issue #8228 + reported by dean bath) + + * /: Blocked revisions 47761 via svnmerge ........ r47761 | file | + 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for + the header file specifically in all cases, not just the existence + of the directory. (issue #8358 reported by mrness) ........ + +2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com> + + * configure, configure.ac: check for pre-1.4 versions of Zaptel and + abort the configure script if found with an appropriate error + message + +2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD + notification optional, in order to avoid a lot of extra database + lookups for all those realtime users out there. + +2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, /: Merged revisions 47750 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov + 2006) | 2 lines Because of the way chan_local is written we + should be extra careful and make sure our callback functions have + a tech_pvt. (issue #8275 reported by mflorell) ........ + + * apps/app_meetme.c: Don't unreference the SLA object if there is + no SLA object in the devicestate callback. (issue #8354 reported + by loloski) + +2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup + + * UPGRADE.txt: Warn users about change in canreinvite + + * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never + authenticated (according to the RFC) - Update docs on + canreinvite. "nonat" is the recommended setting for most users + with phones behind a NAT. + +2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, /: Merged revisions 47711 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov + 2006) | 2 lines Make sure that the pvt structure exists before + trying to do fixup on Local channels. (issue #7937 reported by + mada123, fix by alamantia with mods by me) ........ + +2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL + +2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com> + + * main/channel.c: We need to ensure timelimit stuff is included as + well so warnings get played. (issue #8050 reported by KNK) + +2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com> + + * main/file.c: don't try to call fclose() if fopen() failed + +2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - Improve SIP history - Never send reply to + ACK (again...) + +2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) + | 4 lines ensure that message duration is included in email + notifications for forwarded messages (BE-96, fix by me after + corydon used his clue-bat on me) ensure that duration in the + message metadata is updated if prepending is done during + forwarding (related to BE-96) remove prototype for API call that + does not exist ........ + + * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 + Nov 2006) | 2 lines clear the category's variable tail pointer as + well when variables are detached from it ........ r47688 | + kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 + lines when appending a list of variable to a category, ensure the + tail pointer points to the last variable in the list ........ + r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) + | 2 lines when re-writing the config file, don't repeat the path + if it hasn't changed ........ + + * main/config.c, /: Merged revisions 47682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) + | 2 lines ouch... don't use printf, use ast_log/ast_verbose + ........ + +2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org> + + * main/cli.c: fix longest match search in find_cli. Trunk already + fixed. 1.2 not affected (well, i have no idea, the code is + totally different there). + +2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Send error message when we can't allocate + SIP dialog, possibly due to limitation of file descriptors. + (imported from 1.2) + +2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: If NAT detection is turned on or already detected + then say NAT is active when setting the remote RTP peer when + doing early bridging. (issue #8365 reported by marcelbarbulescu) + +2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com> + + * main/term.c: more formatting cleanup, and avoid running off the + end of the string + +2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Turn notice about unknown RTCP packet type into a + debug message instead. + +2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit + platforms (this variable is an 'int' anyway, comparing it to + 'signed long' is not useful) + +2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 + lines Update copyright information in the ADSI logo blob. + ........ + + * channels/chan_sip.c: Only keep the video RTP structure around if + 1. Video support is enabled and 2. A video codec is enabled on + the dialog + + * funcs/func_uri.c: Small documentation clarification for + URIENCODE. (issue #8294 reported by salaud) + +2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ + prefix did not completely transition app_voicemail when + ODBC_STORAGE is used (reported on IRC by caio1982, not in + bugtracker) + +2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com> + + * apps/app_amd.c: Use LOG_DEBUG to print out the indication that + app_amd is using default settings instead of using LOG_NOTICE. + This stops needless logging of this information under normal + circumstances. (issue #8361 reported by Seb7) + +2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Update documentation to fit the + implementation... + + * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in + retransmission system if it's an OPTION packet from peerpoke + +2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com> + + * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 + lines Initialize global pointers for connection and result to + NULL. (issue #8356 reported by james) ........ + +2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) + | 2 lines Having more than 255 old messages caused corruption in + the new/old count ........ + +2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com> + + * main/config.c: This solves bug 8342, whereby a crash occurs under + certain circumstances while reading a config file with comments-- + a call to CB_ADD shouldn't happen if withcomments is zero + +2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/cli.c, channels/chan_sip.c: Re-enable old deprecated + commands + +2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: - Don't reply to INVITE already replied + to when we get BYE - Declare errmsg as int. Oops. + +2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing + the messed if, but we all forgot to update the regressions. Until + now. + +2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being + found... just confuses users + +2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com> + + * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 + lines When sending an SMS with a user data header properly set + the UDH flag in the first byte. (issue #8347 reported by + hoffmeis) ........ + + * main/cli.c: Free full command string upon unregistering of CLI + command. Backported from revision 47536 from rizzo. + +2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Only produce error message about sip history + once + +2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com> + + * configure, acinclude.m4: AC_PROG_SED is included in autoconf + 2.60, but apparently it is not included in 2.59. So, to maintain + compatability with 2.59 since it is a small change, copy this + macro into acinclude.m4 and rename it to AST_PROG_SED. (issue + #8345) + +2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) + | 2 lines If the execute fails a second time, make sure that we + don't pass back a stale handle ........ + + * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) + | 2 lines Don't play dialtone if the seizing the channel fails + (Bug 7754) ........ + +2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks + DEA!!!) + + * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is + UDPTL in sdp... + + * channels/chan_sip.c: - Don't destroy SIP dialog because of a + failed T.38 re-invite. Wait for a bye. Final response to a + re-invite does not mean that the session dies, only that the + re-invite fails. - Keep RTP active during processing of T.38 + re-invite. If the re-invite fails, RTP needs to remain as before + the re-invite. Issue 8338 - darren1713. Please test. + + * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp + -Add some comments to t.38 code + +2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | + 4 lines Only do the check to determine whether the channel + calling this function is an IAX2 channel when getting the IP + address using the special argument, CURRENTCHANNEL. (issue #8341, + jcovert) ........ + + * Makefile: Add the target "menuconfig" as an alias for the + "menuselect" target. This is just a favor to users so that if you + accidentally type "make menuconfig" instead of "make menuselect", + it still works. (inspired by a comment on IRC from wangster + calling me an "especially devious asterisk developer" for having + it be menuselect instead of menuconfig. :) ) + + * main/term.c: Tweak the formatting of this new function to better + conform to coding guidelines. + +2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com> + + * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo + safe output! + +2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure we don't use 32 bits when we only + need one bit. + +2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: ...and make sure that the dialog is + destroyed, even if we don't get any answer on the bye... This is + the channel that remains dead after the SIP transfer + + * channels/chan_sip.c: Add debug output while trying to trace bug + in bug report + + * channels/chan_sip.c: Make sure we destroy dialog... + + * /, channels/chan_sip.c: Small cleanup of handle_request_invite() + - imported from 1.2 with changes + +2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com> + + * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide + callerid name for switches that bork on it. + +2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart + SDP (alphaque) + +2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org> + + * build_tools/prep_moduledeps: grep -m is not available on BSD, so + use head -1 instead + +2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com> + + * apps/app_chanspy.c: Only split up extension and context if a + value exists. (issue #8332 reported by loloski) + +2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_iax2.c: Discussion of these CLI changes resulted in + more consistency (Bug 8236) + +2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then + removing them should be LOG_NOTICE, not LOG_DEBUG + + * apps/app_queue.c: reflect addition/removal of dynamic queue + members in queue_log, so that people using dialplan replacement + for AgentCallbackLogin can still track login/logout (issue #7736, + reported/patched by whoiswes but this commit was written by me + and covers all three paths for AQM/RQM) + +2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Rip out half implementation of 491 response + support, since it wasn't implemented properly and caused memory + leaks in the case of us getting 491's, which Asterisk actually + sends... Since it is a bit too complicated to fix this, I'll rip + it out of 1.4 and put it on the to-do-list for future releases. + Now, we handle this as congestion, which it really is. Issue + #8331 + + * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. + Thanks fenlander! + +2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com> + + * channels/chan_h323.c: Fix building of chan_h323 by completeing + some structure definitions. (issue #8327 reported by Mithraen) + + * apps/app_voicemail.c: Do conversion in a more easier to read and + working way for \r, \n, and \t. (issue #8324 reported by + johnlange) + +2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c, channels/chan_zap.c, + build_tools/prep_moduledeps: Work around an issue that caused + menuselect to display a bogus description for app_voicemail and + chan_zap. These modules use some preprocessor directives to + determine what it will report to Asterisk as its description. + However, the way we extract this information from the source + files for menuselect is not smart enough to figure this out. + (issue #8326, #8328) + +2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com> + + * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov + 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and + higher as, well, it's apparently going to be removed. This should + make all you FC6 fans happy as your Asterisk will now build + without any mods. ........ + +2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com> + + * main/cli.c: fix tab completion for "core debug channel" and "core + no debug channel" + + * main/cli.c: Fix "core show channel". Also, fix tab completion for + both "core show channel" and "core show channels". + + * main/cli.c: Fix "core debug channel <whatever>". I guess someone + needs to go through and audit every CLI command that changed + number of arguments ... + + * main/asterisk.c: revert the previous change, which actually + modified the deprecated command, "show profile". Now, actually + apply the change to "core show profile". + + * main/asterisk.c: Fix argument parsing for the "core show profile" + CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) + + * main/cli.c: Fix another CLI command, "core show uptime" ... + (issue #8323, reported by johnlange, fixed by myself) + + * main/asterisk.c: fix "core show version" to reflect the new + number of arguments for this CLI command (issue #8316, kshumard) + +2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com> + + * main/channel.c: This update fixes 7531 + + * channels/chan_skinny.c: Committed in behalf of 8190. + +2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com> + + * main/frame.c: the battle over CLI command formats has broken + stuff... + + * channels/chan_sip.c: add simple fix for SDP to report proper + sample rate for G.722 media sessions + +2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com> + + * utils/streamplayer.c: I occasionally get email from users that + are trying to figure out what this does, or due to some + misunderstanding as to what it is supposed to do, can't get it to + work. So, I have added some text here to hopefully explain what + this application does and does not do. + + * channels/chan_gtalk.c: Make this module build again + + * configure, configure.ac, acinclude.m4: Copy the macros from + libtool.m4 to our own acinclude.m4 such that libtool is no longer + required to be installed to be able to generated the configure + script. + +2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) + +2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com> + + * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, + channels/chan_misdn.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_h323.c, + channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, + include/asterisk/stringfields.h, apps/app_voicemail.c, + main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, + channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, + channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to + solve the problem in bug 7506. It's a lot of rework to solve a + fairly small problem... such is life. + +2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c: Make MOH work as it did before in + chan_local, without this then it can go funky when transfers and + MOH are involved. (issue #7671 reported by jmls) + +2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com> + + * configs/musiconhold.conf.sample: clean up sample config, and make + native file playback the more obvious default choice + +2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com> + + * apps/app_voicemail.c: large overhaul to voicemail imap support. + Allows support for more imap servers, also a better + implementation of several parts of the original work. patch + provided by 8033 with major upgrades. + +2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of + continue. + +2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Fixing the attack shield so it doesn't + produce attacks... Issue 8265 - never reply to an ACK + +2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 + Nov 2006) | 5 lines If random order is enabled for files mode + music on hold, set a random initial position, instead of always + starting at the first file, and doing the random operation only + when switching to the next file. (bug reported by John Lange on + the asterisk-dev mailing list) ........ + +2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and + transfer from "john" Thank you! + +2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com> + + * main/cli.c: Fix another bug in "core set debug" ... + + * main/asterisk.c, main/cli.c: Really fix the "core set debug" and + "core set verbose" CLI commands. + + * main/cli.c: fix the "atleast" option to the "core set verbose" + and "core set debug" CLI commands + +2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com> + + * channels/chan_sip.c: This fix introduced via bug 8233 + +2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org> + + * bootstrap.sh: align bootstrap.sh with the version in trunk (needs + to be blocked as it is already in trunk) + + * configure.ac: add proper environment vars to detect modules on + freebsd. (already applied to trunk so it needs to be blocked + there) + +2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, + channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More + changes making the CLI more consistent with "category verb + arguments" (continuation of issue 8236) + + * main/config.c, main/cli.c, main/channel.c, main/manager.c, + channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, + main/http.c, main/file.c, main/logger.c, main/image.c, + res/res_indications.c, main/asterisk.c, res/res_odbc.c, + channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_local.c, main/frame.c, channels/chan_sip.c, + res/res_features.c, channels/chan_agent.c, res/res_crypto.c, + res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: + Reverse change of "show" to "list" and make several other + commands more consistent with "category verb arguments" + +2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Move check for codec translation to + sip_call() instead of in add_sdp. No one bothers with the result + of add_sdp anyway... Yet... + + * channels/chan_sip.c: Disable code for T38 over TCP and RTP since + there's no trace of actual functionality for it :-) + +2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 + Nov 2006) | 3 lines ignore files in a music on hold directory + that begin with '.' (issue #8249, cboie) ........ + +2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com> + + * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix + +2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: don't send INVITE when we have determined + that we can't offer any audio formats due to lack of transcoding + support (or incorrect configuration) + +2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 + lines Repeat after me oej: I will at least make sure my code + compiles before I commit it. ........ + +2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) + +2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com> + + * /, main/callerid.c: Add the missing call to free described in + issue #8268. Also, add a bunch of missing calls to free in + callerid_feed_jp(). + + * main/say.c: fix saying one hundred and two hundred in hebrew + (issue #7810, eldadran) + + * Makefile, configure, codecs/gsm/Makefile, configure.ac, + build_tools/strip_nonapi, makeopts.in: Fixes for + cross-compilation on mips (issue #8058, ywalther, with some + modifications) + + * aclocal.m4, build_tools/menuselect-deps.in, configure, + build_tools/embed_modules.xml, configure.ac: Add a check in the + configure script to determine whether ld is GNU ld or not. This + is needed because module embedding only works for gnu ld. GNU ld + is now listed as a dependency for all of the module embedding + options in menuselect. (issue #8143) + +2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com> + + * channels/chan_gtalk.c: bind address support from bug 8164 + +2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com> + + * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to + accept longer strings or mass confusion and a lot of lost time is + the result + +2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com> + + * main/Makefile: Force poll() emulation for Darwin to always be on. + It's too broken to consider being used. This resolves the console + issue OSX users have been seeing. I would have liked to autoconf + this but I haven't been able to come up with a test case that + works. Que sera. + +2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com> + + * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | + 9 lines soxmix and Asterisk expect different file extensions for + certain formats. This was already handled for the wav49 format. + However, it was not handled for ulaw and alaw. I fixed this in + such a way that using the alternate extensions for ulaw and alaw + will only happen if we know we're calling soxmix, and not a + custom script defined using the MONITOR_EXEC variable. The wav49 + processing was left alone so that external scripts will see no + behavior change. (issue #7550, reported by mnicholson, proposed + patch by junky, committed fix is a bit different) ........ + +2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: It's another round of chan_iax2 fixes! + Should hopefully fix the deadlock issues people have been + reporting. IAXtel now has qualify turned on for 800 peers and it + is handling it fine. + +2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com> + + * main/config.c: Cleanups suggested by Russell. + +2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c: Prevent an infinite loop when config + processing gets to a jitterbuffer option + +2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com> + + * main/translate.c: Fix "core show translation" output. Issue + #8243, patch by Damin. + +2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/translate.h, main/translate.c: add an API so + that translators can activate/deactivate themselves when needed + + * include/asterisk/translate.h, main/translate.c: revert changes + that were the wrong way to address this... proper fix coming + + * main/translate.c: let's set the seen flag early enough to + actually make a difference... + + * include/asterisk/translate.h, main/translate.c: don't re-do setup + operations for translators that can dynamically register + themselves + +2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * /: Blocked revisions 46662 via svnmerge ........ r46662 | + tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines + Move thread-unsafe initializer to the module loading code; add + the corresponding function to the module unload to fix a memory + leak. ........ + +2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net> + + * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue + #8089 - Fix the ENUM support (picking one record by number). + Thanks otmar! + + * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport + when we're supposed to support ;rport. Issue #7473. + + * /, channels/chan_sip.c: If peer fails ACL check, fail peer at + REGISTER + + * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! + +2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com> + + * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the + boot process to ensure it starts after stuff like MySQL (issue + #8253, Alric) + + * /, main/utils.c: Merged revisions 46560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | + 3 lines When handling the case where the hostname is just an IPV4 + numeric address, be sure to set the address type. (issue #8247, + alexr) ........ + + * /, res/res_agi.c: Merged revisions 46557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | + 3 lines fix some copy/paste bugs in the checking of arguments for + the "control stream file" AGI command (issue #8255, mnicholson) + ........ + + * main/translate.c: Add a small tweak to the code that checks to + see whether destination formats are translatable based on the + source format. If we have already determined that there is no + translation path in one direction, don't bother checking the + other direction. + +2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com> + + * main/translate.c: when unregistering a translator, don't rebuild + the translation matrix unless needed when filtering formats out + of an offer, ensure we check for translation ability in both + directions + + * include/asterisk/linkedlists.h: ensure that items removed from a + list are always unlinked from the list (next pointer set to NULL) + +2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com> + + * configure, configure.ac: Don't explicitly link in crypt as it is + not used on some platforms. + + * channels/chan_iax2.c: We need to lock the pvt structure during + retransmission as another worker thread may be doing something as + well. + +2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net> + + * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, + include/asterisk/doxyref.h, channels/chan_sip.c, + main/ast_expr2f.c, include/asterisk/module.h, + formats/format_ogg_vorbis.c, main/app.c, + include/asterisk/channel.h, include/asterisk/lock.h, + include/asterisk/frame.h: Issue #8246 - Doxygen fixes from + kshumard. An extra big thankyou is given to everyone that + contributes to doxygen! THANK YOU! + + * main/rtp.c, /: Bind RTCP to the same IP as RTP + + * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 + redirects (imported from 1.2) + + * /, channels/chan_sip.c: Issue #7608 - Notifications sent with + wrong content-type (imported from 1.2, modified) + + * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that + was reported for trunk, but obviously exists in 1.4 too. + + * channels/chan_sip.c: Restoring the old logic, since working + around it and fixing it seemed too complicated. - The + SIP_OUTGOING flag indicates the direction of the last transaction + in the dialog. - The initreq stores the last request in the + dialog, the request that opened the latest transaction. Please + now retry all the 1.4 bug reports with mixed to/from headers, + tags etc in ACK, BYE, CANCEL. Thanks! + + * channels/chan_sip.c: Accepting a message twice may be + misinterpreted... + + * channels/chan_sip.c: - 183 is not reliable message... - Error + should not have SDP + +2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com> + + * utils/Makefile: Don't build muted on OpenBSD, it is not + supported. + +2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c: move the copy of the default settings to the + global settings back out of process_zap, so that they aren't + overwritten when process_zap is called multiple times + +2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net> + + * contrib/asterisk-ng-doxygen: Put some doxygen pressure on + Christian :-) + +2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com> + + * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, + res/res_musiconhold.c: We should always be using _exit() after a + fork() or vfork() instead of exit(). This is because exit() does + some extra cleanup which in some implementations of vfork(), for + example, can actually modify the state of the parent process, + causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) + + * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell + | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We + should always be using _exit() after a fork() or vfork() instead + of exit(). This is because exit() does some extra cleanup which + in some implementations of vfork(), for example, can actually + modify the state of the parent process, causing very weird bugs + or crashes. (issue #7971, Nick Gavrikov) ........ + + * channels/chan_zap.c: Instead of iterating all of the options once + to look for jitterbuffer options, and then again for everything + else, move the processing of jitterbuffer options into the main + loop so that there are no erroneous messages about ignoring + unknown options. (issue #8226) + +2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 46350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | + 1 line fixed a bug which caused chan_misdn to try to allocate 2 + times the same channel on high load, which then caused + instability of mISDN. removed a useless function from isdn_lib.c + ........ + + * channels/misdn_config.c: fixed not compile issue, which was just + introduced + + * channels/misdn_config.c, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + Merged revisions 46176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | + 1 line added nttimeout option to configure wether we disconnect + calls on NT timeouts or not during an overlapdial session + ........ + +2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com> + + * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 + lines oops - somebody forgot to change this - long ago, probably. + ........ + + * CHANGES: grammar check + +2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net> + + * CHANGES: Corrections to changes (Multiparking is not included) + +2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com> + + * main/translate.c: - If the source has no audio or no video + portion, do not call powerof() to get the format index. - Don't + run through the audio and video loops if there is no audio or + video portion of the source If 0 is passed to powerof, it will + return -1. This value of -1 was then being used as an array index + in these loops, which caused a crash on some systems. Other than + this issue, this code works as we expected it to. If a format is + not in the source, and we have to translation path to it, it is + not offered in the list of acceptable destination formats. (fixes + issue #8231) + +2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com> + + * CHANGES: update to reflect G.722 addition + +2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com> + + * doc/backtrace.txt: update backtrace documentation to reflect + changes in 1.4 (issue #8230, kshumard) + +2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com> + + * main/config.c, main/manager.c: Fix config comment code + preservation code (thanks murf!) + +2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Old todo note - Don't add Contact header on + BYE and Cancel + +2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com> + + * configure.ac: fix error output when checking for openh323 to + refer to openh323 instead of pwlib (issue #8222, misaksen) + +2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Somewhat ugly code to try to fix issue + #7608. Since the problem was not very well defined, the fix is a + bit fuzzy too... Thanks to Luigi for accidentally spotting the + possible problem! + +2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com> + + * apps/app_queue.c: update warning message to include "agi" option + (issue #8225, jmls) + +2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile: use 1.4.3 extra sounds with corrected silence + files + + * sounds/sounds.xml, sounds/Makefile: add support for prebuilt + G.722 prompts and music on hold files + +2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: show settings doesn't produce a list of + similar objects, it should stay a "show" + +2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com> + + * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, + channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, + pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, + main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, + cdr/cdr_custom.c, channels/chan_mgcp.c, + apps/app_parkandannounce.c, apps/app_voicemail.c, + channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, + res/res_adsi.c, main/utils.c, apps/app_ices.c, + pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, + apps/app_getcpeid.c: apparently developers are still not aware + that they should be use ast_copy_string instead of strncpy... fix + up many more users, and fix some bugs in the process + +2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/pbx.c: WaitExten truncates decimals of times to wait, + instead of accepting them (Bug 8208) + +2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com> + + * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, + channels/chan_h323.c, channels/chan_iax2.c, + include/asterisk/frame.h: add passthrough and file format support + for G.722 16KHz audio (issue #5084, original patch by andrew, + updated by mithraen) + + * channels/chan_sip.c, main/translate.c: code zone experiment: + don't offer formats in the outbound INVITE that aren't either + passthrough or translatable + + * main/translate.c: if multiple translators are registered for the + same source/dest combination, ensure that the lowest-cost one is + always inserted earlier in the list + +2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com> + + * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, + #8147) + +2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: We need to initialize our scheduler pthread + condition... yes. + +2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org> + + * main/http.c: merge 45152 don't leak descriptors in http.c + + * channels/chan_sip.c: merge 45966 refer_to_domain potentially + containing options + + * channels/chan_sip.c: merge 46026 improper checks on get_header() + return values + + * channels/chan_sip.c: merge 46045 prevent NULL args to + ast_strdupa() in chan_sip.c + +2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com> + + * Makefile: Restore the ability to remove the firmware directory + without causing the installation to fail (issue #8111) + +2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com> + + * main/translate.c: ensure that the translation matrix is properly + lock-protected every place it is used + + * include/asterisk/translate.h, main/translate.c: add an API call + to allow channel drivers to determine which media formats are + compatible (passthrough or transcode) with the format an existing + channel is already using + + * doc/imapstorage.txt: simplify and correct voicemail IMAP storage + build instructions + +2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * main/channel.c: Pass through a frame if we don't know what it is, + rather than trying to pass a NULL, which will segfault a channel + driver (Bug 8149) + +2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com> + + * utils/muted.c, utils/ael_main.c: In muted.c, check the return + value of strdup. In ael_main.c, check the return value of calloc. + (issue #8157) In passing fix a few minor bugs in ael_main.c. The + last argument to strncpy() was a hard-coded 100, where it should + have been 99. I changed this to use sizeof() - 1. + + * apps/app_meetme.c: Fix the descriptions of some of the + MeetMeAdmin options (issue #8098, mflorell) + + * res/res_jabber.c: don't crash when an incoming message has no + "from" (issue #8205, jmls) + +2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com> + + * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 + lines Don't leak memory mmmk? ........ + +2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 + Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and + couldn't be initialized it would cause a segfault after 'reload'. + Reported by Drew/Matt thx. ........ + +2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com> + + * res/res_monitor.c: Add a couple missing unregistrations of + manager actions and remove duplicate unregistrations of + applications. (issue #8194, jmls) + +2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com> + + * main/loader.c: Don't use promotion on Darwin because it doesn't + seem to work quite right in all cases, this should solve the + unresolved symbol issue people have been seeing. + + * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get + installed in the proper location (reported on asterisk-dev + mailing list) + +2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Let's understand SIP: - REFER can create + dialog, Asterisk does not support it yet - NOTIFY can create + dialog in Asterisk's implementation (voicemail) even though we + don't support the server side of it. In this case, the standard + is a side issue ;-) - Added extened functionality for unsupported + methods (PING, PUBLISH) so we don't create PVT's for those + either. Russellb needs to judge what to do with this in 1.2, but + I think the current implementation n 1.2 is a bug since we're + sending bad replies to NOTIFY and REFER outside of dialogs + +2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com> + + * res/res_jabber.c: Let's remember to unregister JabberStatus too + (issue #8184 reported by jmls) + + * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct + 2006) | 2 lines Respect language selection when seeing if the + file exists (issue #8178 reported by mnicholson) ........ + + * channels/chan_sip.c: If the jitterbuffer is forced on then we + can't partially bridge (reported by wangster on #asterisk-dev) + +2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Don't leak the actual thread-specific + sip_pvt struct + +2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: don't leak memory when a chan_sip thread is + destroyed that has a thread-local temp_pvt allocated + +2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com> + + * main/asterisk.c: Don't modify things if we are using vfork as + this is very bad and may cause unexpected behavior (issue #7970 + reported by Nick Gavrikov) + +2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: remove duplicate declarations + +2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org> + + * main/http.c: merge from trunk: move ast_variables_destroy() to a + better place in handle_uri() to avoid leaking memory on non + existing files. + +2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Don't segfault if you're using a channel driver that + doesn't turn RTCP on + +2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com> + + * main/channel.c: Don't attempt to access private data members of + the pthread_mutex_t object, because this does not work on all + linux systems. Instead, just access the reentrancy field in the + ast_mutex_info struct when DEBUG_THREADS is enabled. If + DEBUG_CHANNEL_LOCKS is enabled, the developer probably has + DEBUG_THREADS on as well. (issue #8139, me) + + * configs/sip_notify.conf.sample: update entry to reboot a snom + phone (issue #7850, pnlarsson) + +2006-10-17 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.4.0-beta3 released. + +2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/stringfields.h, main/ast_expr2.c, + main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: + optimize the 'quick response' code a bit more... no more malloc() + or memset() for each response expand stringfields API a bit to + allow reusing the stringfield pool on a structure when needed, + and remove some unnecessary code when the structure was being + freed + +2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Don't create a "real" pvt structure for + requests that shouldn't be able to create one. Instead use a + temporary pvt and fill it with enough information so we can send + a reply. + +2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net> + + * configs/sip.conf.sample: Adding information about Marks + direct-RTP hack to the docs... + +2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com> + + * LICENSE: provide licensing language for IAXy firmware file + +2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new + directed pickup (BE-85). + +2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net> + + * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for + your support! + + * channels/chan_sip.c: Don't destroy dialog for unexpected REFER + response... + +2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com> + + * funcs/func_rand.c: update the doc string for both AEL and + extensions.conf users. + +2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com> + + * main/acl.c don't drop the entire permit/deny list when an attempt + is made to add an invalid entry (BE-92) + +2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com> + + * res/res_speech.c: Clear the quiet flag too since we are + restarting a recognition again (reported on -dev by Stephan + Edelman) + + * res/res_speech.c: Check return value from engine in case of + failure (ie: out of licenses) (reported on -dev mailing list) + +2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-vtest17 (added), + pbx/ael/ael-test/ael-vtest17/extensions.ael (added), + pbx/ael/ael-test/ael-vtest17 (added), + pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in + this release via these changes + +2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: avoiding warning, fixing potential bug + +2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com> + + * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, + codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, + codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, + codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, + codecs/lpc10/synths.c, codecs/lpc10/preemp.c, + codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, + codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, + codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, + codecs/lpc10/lpcini.c, codecs/lpc10/random.c, + codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, + codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, + codecs/lpc10/analys.c, codecs/lpc10/onset.c, + codecs/lpc10/energy.c, codecs/lpc10/deemp.c, + codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, + codecs/lpc10/median.c, codecs/lpc10/encode.c, + codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, + codecs/lpc10/invert.c: And file said... let the compiler warnings + STOP! + + * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 + reported by mnicholson) + + * apps/app_playback.c: Move say.conf existence check to do_say + function since it is called from multiple places (issue #8144 + reported by kshumard) + +2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if + we have multiple bindings (reported on asterisk-dev) + +2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Complete merging in RPID screen changes + (issue #8101 reported by hristo, patch by oej in revision 44757) + + * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add + the background refresh item back into the scheduler if enabled + since it is deleted during reload. (issue #8142 reported by + p_lindheimer) + +2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/utils.c: use a configure script test for PMTU discovery + control instead of just assuming it's available on Linux + +2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some + echocandisable issues when bridged. this caused a kernel panic + sometimes.. also some minor formatting fixes + + * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause + got a wrong isdn cause at RELEASE_COMPLETE + +2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: merge formatting and minor code + simplifications from trunk + +2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com> + + * channels/chan_gtalk.c: fix for bug 7764. + +2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: we can only send one 'a=ptime' attribute per + media session, not one for each format + + * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, + main/utils.c: ensure that IAX2 and SIP sockets allow UDP + fragmentation when running on Linux (thanks to Brian Candler on + the asterisk-dev list for the tip) + +2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com> + + * main/manager.c: fix a silly typo in a comment that I saw while + reading the commit list + +2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com> + + * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue + #8135 reported by ssokol) + +2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com> + + * main/manager.c: append_event must be called while holding the + session lock + +2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com> + + * res/res_jabber.c: change some debug output to use LOG_DEBUG + instead of verbose output + +2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com> + + * main/db1-ast/Makefile: These are already set by the parent + Makefile.. There is no need to have this here (it doesn't + actually work anyways). + +2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c: removed warning because of missing + prototype declaration + +2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Do not set default/global values in the + variable declaration, set it in reload_config() + +2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Move some stuff around so that a NOTIFY + dialog won't hang around until the end of the world under certain + circumstances + +2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz> + + * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: + CHANNEL() function sometime mix parameter and value + +2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * funcs/func_logic.c: Lost of a bit of logic when this was + simplified between 1.2 and 1.4 (Bug 8117) + +2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Bail out if we have no refer structure and + we get a refer response + +2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: more merge from trunk (comments and change a + static function name) + +2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Only set DTMF information if an RTP + structure exists + +2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added + support of dynamically enabling hdlc on bchannels + +2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: whitespace changes related to previous + commit + + * channels/chan_sip.c: merge a few code simplifications that have + gone into trunk during last week, to reduce differences between + the two branches and make porting fixes easier. + +2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Fix a problem where phones that go + "missing" never got unregistered. Issue #8067, reported by pj, + patch by Anthony LaMantia (with minor whitespace modifications) + +2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid + the deadlock + + * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup + (issue #8115 reported by vazir) + +2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: do not dereference p if we + know it is NULL + +2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate + caller's transfer capability too + +2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: put common code in a + function to avoid repetitions. + + * channels/chan_sip.c: remove hardwired usage of 5060, use + DEFAULT_SIP_PORT instead + + * channels/chan_sip.c: option_debug checking + before printing to debug channel. + + * channels/chan_sip.c: backport simplifications on sip_register, + usage of ast_set2_flag(), and fixes to the handling of failed + module loading. + + * channels/chan_sip.c: improve and document function + get_in_brackets(), introducing a helper function + find_closing_quote() of more general use. + +2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/linkedlists.h: ensure that mutex locks inside + list heads are initialized properly on platforms that require + constructor initialization (issue #8029, patch from timrobbins) + + * CHANGES: remove Jingle as per mog + +2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Remove the seqno check for RFC2833, the handler is + smart enough to not need it. + +2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com> + + * CHANGES: various cleanups + +2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: When the sequence number rolls over then reset the + recorded sequence number for DTMF (issue #8106 reported by + bungalow) + + * main/file.c: Even more frames to treat as though the remote side + disappeared (issue #8097 reported by eldadran) + +2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org> + + * main/manager.c, main/http.c: make sure sockets are blocking when + they should be blocking. + +2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: fixed segfault which happens during + hold/transfer action + + * channels/chan_misdn.c: if INFORMATION Message come with keypad + instead of called party number, we just use the keypad as called + party number. + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + added the option 'reject_cause' to make it possible to set + the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, + which is automatically rejected because chan_misdn does not + support that kind of callwaiting. Therefore chan_misdn supports + now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc + now gets the info if the requested channel is incoming or + outgoing to make the 3. channel possible + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, + removed a useless bc field, added setting of frame.delivery fields, + some minor code cleanups + +2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com> + + * main/file.c: Treat busy control frames as hangup in the file streaming + core (issue #8097 reported by eldadran) + +2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. + Many thanks to Doug! + +2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite + hanging by a thread if the other side is already setup with T.38 + +2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com> + + * main/app.c: don't segfault when an argument without a close + parenthesis is found stop parsing as soon as that situation + occurs + +2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com> + + * CHANGES: I put the accumulated changes from the commit logs and + inspection, into CHANGES. Hope everyone approves! + + * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the + install process sticks muted.conf in /etc/asterisk, so that's + where muted should look for it, right? + +2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Don't totally bail out if T.38 was + negotiated + +2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: fix Polycom presence notification again + +2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org> + + * utils/Makefile: as far as i can tell astman only uses newt... + + * Makefile: put linker flags in ASTLDFLAGS where they belong + +2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE + requests add workaround for new Polycom firmware SUBSCRIBE + requests (bug is known to exist in 2.0.1 firmware) + + * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually + work + +2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, + pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, + pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, + pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, + pbx/ael/ael-test/ael-test16/extensions.ael (added), + pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, + pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, + pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, + pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the + problems reported in bug 8090 + +2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, + main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, + main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, + include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, + main/devicestate.c, main/utils.c, res/res_musiconhold.c, + channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update + thread creation code a bit reduce standard thread stack size + slightly to allow the pthreads library to allocate the stack+data + and not overflow a power-of-2 allocation in the kernel and waste + memory/address space add a new stack size for 'background' + threads (those that don't handle PBX calls) when LOW_MEMORY is + defined + +2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com> + + * configs/muted.conf.sample: I've been meaning to add some + explanation about muted... here it is + + * configs/manager.conf.sample: CLI reverbification update to this + config file + + * apps/app_macro.c: In response to bug 7776, a Warning has been + added to the doc string for Macro(). + +2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com> + + * main/asterisk.c, main/loader.c, main/term.c, Makefile, + include/asterisk.h: ensure that local include files are always + used avoid a duplicate function name (term_init()) + +2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com> + + * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing + client without resource. + +2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_queue.c: fix a logic error in my previous fix to the queue + reload code + +2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Change default presentation indicator + to "user provided not screened" if octet 3a missed in + CallingPartyNumber IE + +2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Use VideoSupport instead so it is considered + a valid XML attribute name. (issue #8075 reported by renemendoza) + +2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Fix preparation of type and + presentation of calling number + +2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com> + + * doc/jingle.txt, channels/chan_jingle.c (removed), + include/asterisk/jabber.h, configs/jingle.conf.sample (removed), + res/res_jabber.c: updated res_jabber for even better component + support, soon will be jep-0100 compliant. also removed + chan_jingle and infromed info from jingle.txt, chan_gtalk still + works and should be used in this version. + +2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Change the fd on the I/O context in case it + changed during the reload, which is indeed possible. (issue #7943 + reported by eclubb) + + * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN + instead of hardcoding the path for the error message (issue #7942 + reported by eclubb) + +2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz> + + * configs/users.conf.sample, pbx/pbx_config.c: Missed part of + userconf functionality for chan_h323 + +2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com> + + * main/io.c: Shrink when current_ioc is unused. It is set to -1 when + unused, not 0. (issue #7941 reported by eclubb) + +2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz> + + * doc/realtime.txt: Typo fix + + * channels/chan_h323.c: Optimization of oh323_indicate(): less + locks - less problems, plus single exit point + +2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com> + + * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when + you're not talking about a channel :) + +2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_h323.c: Do not simulate any audio tones if we got + PROGRESS message + +2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com> + + * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to + be empty. The cause is that since ASTDATADIR is explicitly + exported using "export ASTDATADIR" at the top of the Makefile, + make no longer considers the variable "undefined", so the + Makefile can't use ?= to set ASTDATADIR if not yet set. (issue + #8063, reported by akohlsmith, fixed by me) + + * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" + option in the sample queues.conf (issue #8065, adamg) + +2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_sip.c: sync with trunk - move variable declarations + to the beginning of a block. + +2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz> + + * main/rtp.c: Allow one-way RTP streams (device->Asterisk) + +2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org> + + * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent + build problems: - with AST_DEVMODE, building codecs/lpc10 fails + because of lots of warnings, and the configure step in editline + fails as well. Fix this by removing the -Werror in these steps. - + on FreeBSD (but probably on other platforms as well), the final + link of asterisk fails because AST_LIBS was not exported to the + subdirs Makefiles. Add a proper fix in the top-level Makefile (a + possible alternative way is to add "export AST_LIBS" near the + beginning of the file). With this fix, i believe that some of the + platform-specific conditionals in main/Makefile are redundant + (because they should be already dealt with in the top level + Makefile) but i don't have a platform to check. Merging to head + will happen in a moment. + +2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment + of previous fix: Issue #7928 - Don't send both 404 and 503. Fix + by phsultan with a small fix by me, myself or I. Thanks, + Philippe! (This was caused by my changes to the transaction + handling) + + * channels/chan_sip.c: Found some buggy SIP clients (phones Planet + VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which + sends ACK not on OK message only (when remote party answers) but + on RINGING message too, so when we send 200 OK message, we get + unidentified ACK message (because INVITE acknowledged on RINGING + message already), so 200 OK retransmits within its retransmission + interval then call gets dropped. If someone else knows how to + provide workaround for such cases, please, fix it in correct way. + Thanks to ssh from #asteriskru for provide access to his box to + study and fix this case. + +2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com> + + * agi, utils: ignore temporary files made by the Makefiles during a + build + + * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, + codecs/Makefile, utils/Makefile, configure, + build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, + Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, + pbx/Makefile, res/Makefile, channels/Makefile: fix a few build + system bugs, and convert Makefiles to be compatible with GNU make + 3.80 + +2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com> + + * main/asterisk.c, main/cli.c: Fix a bug with the removal of + 'atleast' argument to 'core verbose' and 'core debug'. Add that + argument back in. + +2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more + carefully when no CallingNumber IE available + + * channels/h323/ast_h323.cxx: Fake display name by called number on + incoming calls (until passing connected number/connected name is + not implemented) + + * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add + includes + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly + pass TON/PRESENTATION information - original + H323Connection::SendSignalSetup() destroys Q.931 fields. + +2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com> + + * main/Makefile: yet another place where we were not using the + correct CFLAGS by default + + * main/Makefile: missed one conversion to ASTCFLAGS + +2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass + TON/PRESENTATION information too + +2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com> + + * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, + main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, + Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse + CFLAGS and LDFLAGS for build of Asterisk components, because they + are also then used for non-Asterisk components (like menuselect); + use our own variables instead + + * configure, configure.ac: support --without-curl in configure + script + + * Makefile.rules: another cross-compile fix + + * Makefile: a couple more environment settings that can't leak into + the menuselect build + + * main/cli.c: proper fix for ast_group_t change + + * include/asterisk/lock.h: eliminate compiler warning when + DEBUG_CHANNEL_LOCKS is enabled and users of this header file + don't also include channel.h + +2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com> + + * apps/app_queue.c: Fix incorrect argument order for member names, + on persisted members. Issue 8047, patch by jmls. + +2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com> + + * apps/app_playback.c, res/res_monitor.c, + include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, + channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/udptl.c, main/frame.c, funcs/func_timeout.c, + channels/chan_sip.c, apps/app_festival.c, + channels/iax2-provision.c, apps/app_alarmreceiver.c, + res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: + Put in missing \ns on the end of ast_logs (issue #7936 reported + by wojtekka) + +2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_queue.c: fix buggy (and overly complex) loop used during reload + of app_queue for static member list updating + +2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Extend call establishment timeout + +2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: Make sure the pvt exists before accessing + it again as it may have gone away (issue #7562 reported by Seb7 + and issue #7939 reported by sorg) + + * main/cli.c: Warning be gone! + +2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com> + + * apps/app_queue.c: app_queue is comparing the device names incorrectly + while checking their statuses. It's internal list of interfaces + includes the dial string, while the argument passed to this + function does not have the dial string (/n for a local channel). + This causes it to ignore the device state changes because it + thinks it belongs to none of its members. (#8040 reported and + patch by tim_ringenbach) + +2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com> + + * apps/app_meetme.c: Stop the stream after waitstream returns so that our + formats get restored. (issue #7370 reported by kryptolus) + +2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Fix compiler warning + +2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com> + + * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - + tim_ringenbach reported and patched) + + * apps/app_queue.c: Autopause not working for queue members. (#8042 + - jmls reported and patch) + +2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force + remote side to start media on outgoing PROGRESS message + + * include/asterisk/compiler.h: Put attribute tag at correct place + +2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com> + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed a bug which led to chan_list zombies, + when the call could not be properly established in misdn_call. + also removed the ACK_HDLC stuff which is not really needed. + +2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/ast_h323.cxx: Do not open transmit channel until + TCS is received + + * main/file.c: Don't warn on HOLD/UNHOLD control frames + + * main/file.c: Don't treat unknown control frames as voice + +2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Avoid inability to lock directory log message by + creating the directory ahead of time. (Issue 7631) + +2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com> + + * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS + not being set under certain circumstances. Fix a minor issue, to + make it use the filenames that were parsed, instead of the entire + argument string. Fix Background() to return -1 like Playback(), + if no args are specified. + +2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Compensate for out of order packets better if RFC2833 + compensation is turned on. + + * channels/chan_iax2.c: Get rid of two functions from a time now + past (we THINK these are from pre-recursive lock time) that may + be contributing to two open issues on the bug tracker (7562/7939) + and that has the potential to just make bad things happen if the + timing is right. + +2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com> + + * main/channel.c,res/res_features.c: Fix a problem that occurred if + a user entered a digit + that matched a bridge feature that was configured using multiple + digits, and the digit that was pressed timed out in the feature + digit timeout period. For example, if blind transfer is + configured as '##', and a user presses just '#'. In this + situation, the call would lock up and no longer pass any frames. + (issue #7977 reported by festr, and issue #7982 reported by + michaels and valuable input provided by mneuhauser and kuj. Fixed + by me, with testing help and peer review from Joshua Colp). There + are a couple of issues involved in this fix: 1) When + ast_generic_bridge determines that there has been a timeout, it + returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets + this result, it calls ast_generic_bridge over again with the same + timestamp for the next event. This results in an endless loop of + nothing until the call is terminated. This is resolved by simply + changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it + sees a timeout. 2) I also changed ast_channel_bridge such that if + in the process of calculating the time until the next event, it + knows a timeout has already occured, to immediately return + AST_BRIDGE_COMPLETE instead of attempting to bridge the channels + anyway. 3) In the process of testing the previous two changes, I + ran into a problem in res_features where ast_channel_bridge would + return because it determined that there was a timeout. However, + ast_bridge_call in res_features would then determine by its own + calculation that there was still 1 ms before the timeout really + occurs. It would then proceed, and since the bridge broke out and + did *not* return a frame, it interpreted this as the call was + over and hung up the channels. The reason for this was because + ast_bridge_call in res_features and ast_channel_bridge in + channel.c were using different times for their calculations. + channel.c uses the start_time on the bridge config, which is the + time that the feature digit was recieved. However, res_features + had another time, 'start', which was set right before calling + ast_channel_bridge. 'start' will always be slightly after + start_time in the bridge config, and sometimes enough to round up + to one ms. This is fixed by making ast_bridge_call use the same + time as ast_channel_bridge for the timeout calculation. ........ + +2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn + versioning, since Asterisk has it's own + +2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Make rfc2833compensate a global option. + +2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: Backport revision 43754 from the trunk, + which removes an unused buffer from mm_login to close bug 8038, + as well as addresses some formatting and coding guidelines issues + in passing. Originally, I did not commit this to 1.4 since it is + not necessarily fixing a bug. However, since the IMAP storage + code is brand new, I decided it would be better to make the + change here as well, in case someone has to work on this code to + address issues in the very near future. I don't want to make + unnecessary merge problems going to the trunk. + +2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com> + + * configs/extensions.ael.sample: This change to extensions.ael was + to fix bug 8031; the install scripts are causing it to be copied + to /etc/asterisk/extensions.ael, and because it is a fairly + direct conversion of the original extensions.conf, the macro and + context names clash with the existing extensions.conf. So, I put + an ael- in front of all macros and contexts, and checked every + goto and macro call. Also, this file compiles under aelparse. + +2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com> + + * main/asterisk.c: Back in revision 4798, this message was changed from + using ast_cli() to directly calling write(). During this change, + checking if this was a remote console was removed. This caused + this message about using "exit" or "quit" to exit an Asterisk + console to come up in times where it did not make sense. This + change restores the check to see if this is a remote console + before printing the message. (fixes BE-65) + +2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com> + + * .cleancount, main/cli.c, channels/chan_sip.c, + include/asterisk/channel.h: Use proper type to represent the group variable + (issue #8025 reported by makoto) + +2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Add missing newline character in the warning + message about deprecated TOS values in configuration. + + * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain + mailbox definitions, don't introduce a length limit on the + definition by using a 256 byte temporary storage buffer. Instead, + make the temporary buffer just as big as it needs to be to hold + the entire mailbox definition. (fixes BE-68) + +2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c: Strip options off the argument passed for + devicestate in chan_local. (issue #8034 reported by pcardozo) + + * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight + overhaul of the whisper support. 1. We need to duplicate the + frame from ast_translate 2. We need to ensure we always have + signed linear coming in for signed linear combining. 3. We need + to ensure we are always feeding signed linear out. 4. Properly + store and restore write format when beeping on the channel we are + whispering on. 5. Properly discontinue the stream on the channel + for the beep. (issue #8019 reported by timkelly1980) + +2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile: update to use 1.4.3 core sounds, with corrected + beep/beeperr/tt-monkeys files + +2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com> + + * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by + Dan Austin. Maximum values were incorrect, which is why this is + being put in 1.4 + + * channels/chan_skinny.c: Add proper codec support to chan_skinny. + Works with at least ulaw, alaw, and g729a. This is technically a + "new feature", but there are justifications for it. I found a bug + with the recent rtp packetization changes, which caused the media + setup to fail under certain circumstances, particularly when + using allow=all, or having no allow= statements (globally or on + the device). I could have either removed the rtp packetization + features, or I could add proper codec support (which, without, I + think most people would consider to be a bug anyways). + +2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_voicemail.c: Should have moved these lines up in the + merge, instead of removing them + + * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) + delete=yes was ignored 2) maxmessages was ignored + +2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, + channels/h323/cisco-h225.asn: Fix ASN1 description of + non-standard Cisco extensions + + * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport + changes of trunk: 1) r43540: Avoid possible deadlock on channel + destruction 2) r43590: Disable fastStart if requested by remote + side + +2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com> + + * sounds/Makefile: One more fix for sounds installation - this time + for portability. Reported to asterisk-dev mailing list. + +2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com> + + * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from + crashing if trying to play an OGG moh file. + +2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz> + + * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, + channels/chan_h323.c: Merged revisions 43472,43495 from trunk + +2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com> + + * channels/iax2-provision.c: Fix a CLI command registration issue + where an erroneous message claiming that "iax2 show provisioning" + was already registered. This was because this command was + registering itself as both the command, as well as the command it + is deprecating. (issue #8022, reported by bjweeks, fixed by + myself) + + * channels/chan_iax2.c:Check to see if the channel that is activating the + IAXPEER function is actually an IAX2 channel before proceeding to + process it to avoid crashing. (issue #8017, reported by admott, + fixed by myself) + +2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile: don't output the 'build complete' message when the + target being run is already going to do an installation + +2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded + properly. Remove reload support, since it doesn't + actually...work. + +2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com> + + * pbx/pbx_ael.c: This commits a change to return + MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all + goes well for bug 8004 + + * pbx/pbx_ael.c: If the extensions.ael file not found, or + unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. + +2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com> + + * main/cli.c: Make sure we explicitly set the CLI command to not be + deprecated, if it isn't. + +2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com> + + * sounds/Makefile: use rebuilt extra sounds + + * main/channel.c: all the Linux systems I have don't use + '__m_count' for this field, so I don't know where this came + from... + +2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com> + + * include/asterisk/threadstorage.h: backport the compatability fix + to use attribute_malloc instaed of __attribute__ ((malloc)) + + * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN + could not be configured (issue #8006, Mithraen) + + * main/frame.c: Suppress a compiler warning about the use of a + potentially uninitialized variable. It couldn't actually happen, + though. + +2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: First shot at unload_module in + chan_skinny.. More to come. + +2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com> + + * include/asterisk/jabber.h, channels/chan_gtalk.c, + res/res_jabber.c: updates for better compontent support + +2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we + actually documented how the new features in res_odbc actually + work. (Oops) + +2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com> + + * channels/chan_oss.c: Some more clean up in the load function for + chan_oss (issue #8002 reported by Mithraen with minor mods by + moi) + + * channels/chan_mgcp.c: Clean up chan_mgcp's module load function + (issue #8001 reported by Mithraen with mods by moi) + +2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com> + + * main/Makefile, build_tools/strip_nonapi (added): add another + attempt to strip non-API symbols from the final binary... script + will need to be extended to work on non-Linux systems + +2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> + + * apps/app_url.c: Fix documentation to reflect how Url() really + works + + * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates + +2006-09-21 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.4.0-beta2 released. + +2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com> + + * main/Makefile: remove this change... it requires binutils 2.17 + +2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com> + + * build_tools/make_version: fix minor typo in the way version is + handled + +2006-09-20 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.4.0-beta1 released. |