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2009-02-18Merged revisions 176760 via svnmerge from sruffell1-80/+321
https://origsvn.digium.com/svn/asterisk/trunk ........ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176805 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Backing out an improper merge from trunk.sruffell1-321/+80
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176804 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176760 via svnmerge from sruffell1-80/+321
https://origsvn.digium.com/svn/asterisk/trunk ........ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176803 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176771 via svnmergerussell0-0/+0
........ r176771 | russell | 2009-02-17 16:52:43 -0600 (Tue, 17 Feb 2009) | 2 lines Remove a dependency that no longer exists. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176772 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176708 via svnmerge from jpeeler3-14/+33
https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176710 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176705 via svnmerge from dhubbard1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176709 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176697 via svnmerge from mmichelson1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176698 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176666 via svnmergerussell0-0/+0
........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176670 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176642 via svnmerge from tilghman1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176643 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176639 via svnmergerussell0-0/+0
........ r176639 | russell | 2009-02-17 15:04:08 -0600 (Tue, 17 Feb 2009) | 9 lines Significantly improve scheduler performance under high load. This patch changes the scheduler to use a max-heap to store pending scheduler entries instead of a fully sorted doubly linked list. When the number of entries in the scheduler gets large, this will perform much better. For much more detailed information on this change, see the review request. Review: http://reviewboard.digium.com/r/160/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176640 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176635 via svnmergerussell0-0/+0
........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 Feb 2009) | 4 lines Add a test module for the heap implementation. Review: http://reviewboard.digium.com/r/160/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176636 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176632 via svnmergerussell0-0/+0
........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176633 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176627 via svnmergerussell0-0/+0
........ r176627 | russell | 2009-02-17 14:41:24 -0600 (Tue, 17 Feb 2009) | 37 lines Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176628 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Blocked revisions 176557 via svnmergerussell0-0/+0
........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176558 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Oops, merge broke 1.6.0.tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17In 1.6.0, the tablename is stored in a variable.tilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176502 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176459 via svnmerge from tilghman1-6/+29
https://origsvn.digium.com/svn/asterisk/trunk ................ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176460 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176355 via svnmerge from dvossel1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176359 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Blocked revisions 176356 via svnmergekpfleming0-0/+0
........ r176356 | kpfleming | 2009-02-16 17:37:37 -0600 (Mon, 16 Feb 2009) | 3 lines add support for Siren7 and Siren14 flavors of prompts and music on hold ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176357 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176255 via svnmerge from kpfleming2-24/+44
https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176258 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176253 via svnmerge from mmichelson1-14/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176256 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176248 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176174 via svnmerge from mmichelson1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176175 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Blocked revisions 176100 via svnmergerussell0-0/+0
........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines Remove chan_features. Review: http://reviewboard.digium.com/r/161/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176101 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Eliminate mention of a variable which is only available in trunk.tilghman1-6/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176030 via svnmerge from file1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176031 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 175952 via svnmerge from mvanbaak4-8/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176022 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Blocked revisions 175983 via svnmergerussell0-0/+0
........ r175983 | russell | 2009-02-15 20:54:42 -0600 (Sun, 15 Feb 2009) | 2 lines Make the causes array static, and remove the type name as it is not needed. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175984 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Blocked revisions 175882 via svnmergerussell0-0/+0
........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175887 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Blocked revisions 175829 via svnmergerussell0-0/+0
........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175830 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175827 via svnmerge from oej1-4/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r175827 | oej | 2009-02-15 21:39:55 +0100 (Sön, 15 Feb 2009) | 10 lines Merged revisions 175825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175828 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175801 via svnmerge from oej1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r175801 | oej | 2009-02-15 21:22:12 +0100 (Sön, 15 Feb 2009) | 10 lines Merged revisions 175792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines Disable format_ilbc.so by default, like codec_ilbc.so ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175816 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175623,175636 via svnmergerussell0-0/+0
........ r175623 | russell | 2009-02-13 14:23:39 -0600 (Fri, 13 Feb 2009) | 1 line add missing </para> ........ r175636 | russell | 2009-02-13 14:26:49 -0600 (Fri, 13 Feb 2009) | 1 line fix a few more XML documentation problems ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175591 via svnmergemmichelson0-0/+0
................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175592 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Merged revisions 175549 via svnmerge from file1-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175550 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175512 via svnmergekpfleming0-0/+0
........ r175512 | kpfleming | 2009-02-13 07:41:52 -0600 (Fri, 13 Feb 2009) | 3 lines document G.722.1/.1C support ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175514 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175508 via svnmergekpfleming0-0/+0
........ r175508 | kpfleming | 2009-02-13 07:35:24 -0600 (Fri, 13 Feb 2009) | 15 lines Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175475 via svnmergedhubbard0-0/+0
........ r175475 | dhubbard | 2009-02-12 22:22:35 -0600 (Thu, 12 Feb 2009) | 1 line add 'faxbuffers' configuration option information to CHANGES ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175476 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175411 via svnmergedhubbard0-0/+0
........ r175411 | dhubbard | 2009-02-12 18:13:38 -0600 (Thu, 12 Feb 2009) | 13 lines Add dynamic fax buffer configuration option to chan_dahdi.conf When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175472 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175368 via svnmerge from russell1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Blocked revisions 175344 via svnmergedvossel0-0/+0
........ r175344 | dvossel | 2009-02-12 15:27:11 -0600 (Thu, 12 Feb 2009) | 10 lines Adds force encryption option to iax.conf This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175366 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175334 via svnmerge from tilghman1-23/+51
https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175347 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Fix mistake in merging conflict from 175299.jpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175298 via svnmerge from jpeeler1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175299 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175295 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175296 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175255 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175256 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Blocked revisions 175250 via svnmergekpfleming0-0/+0
........ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line correct warning message to not refer specifically to DAHDI ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175252 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175188 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175189 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175125 via svnmerge from russell1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175126 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175121 via svnmerge from mmichelson2-2/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175122 f38db490-d61c-443f-a65b-d21fe96a405b
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