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Author	SHA1	Message	Date
Jeff Peeler	46db811169	Merged revisions 176708 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	24ef2a922e	Merged revisions 176705 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ddee1048c7	Merged revisions 176697 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	603e58ad3a	Blocked revisions 176666 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f8b942b064	Merged revisions 176642 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dab9950656	Blocked revisions 176639 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8c08a905f9	Blocked revisions 176635 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ecd8f84421	Blocked revisions 176632 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1a121a8737	Blocked revisions 176627 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fabeae2b4a	Blocked revisions 176557 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	dce3b59e28	Oops, merge broke 1.6.0. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0ab34a15f5	In 1.6.0, the tablename is stored in a variable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d019ac6ef4	Merged revisions 176459 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f62cf2d8e3	Merged revisions 176355 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f64c4404b3	Blocked revisions 176356 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f3ae0e00a7	Merged revisions 176255 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d504f89a9d	Merged revisions 176253 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	af1b475948	Merged revisions 176248 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1999cdee6c	Merged revisions 176174 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	968aada889	Blocked revisions 176100 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c114192b13	Eliminate mention of a variable which is only available in trunk. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d86ea77d16	Merged revisions 176030 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	25c01347d8	Merged revisions 175952 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	55f3396445	Blocked revisions 175983 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	854d364ce9	Blocked revisions 175882 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	abf8b910f9	Blocked revisions 175829 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	cd8a69a565	Merged revisions 175827 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	fc681ee891	Merged revisions 175801 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	36cbd1bf2f	Blocked revisions 175623,175636 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	aa60558799	Blocked revisions 175591 via svnmerge ... ................ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	496e168b87	Merged revisions 175549 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	351eab03b2	Blocked revisions 175512 via svnmerge ... ........ r175512 | kpfleming | 2009-02-13 07:41:52 -0600 (Fri, 13 Feb 2009) | 3 lines document G.722.1/.1C support ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b0722c20ca	Blocked revisions 175508 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	89c50d282d	Blocked revisions 175475 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	145fff8f3b	Blocked revisions 175411 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	48ade8a53e	Merged revisions 175368 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4ddeba5e16	Blocked revisions 175344 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	96a87efb7f	Merged revisions 175334 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8008dca29d	Fix mistake in merging conflict from 175299. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	0a72cfe440	Merged revisions 175298 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	869e8bc417	Merged revisions 175295 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ecd3f17267	Merged revisions 175255 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2a60c89b53	Blocked revisions 175250 via svnmerge ... ........ 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	17df1c9d13	Merged revisions 175188 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d79cc1e799	Merged revisions 175125 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	90ef4eb33e	Merged revisions 175121 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	4c8559e441	Merged revisions 175089 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	e28edbe89c	Merged revisions 175058 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b8356f9a94	Merged revisions 174948 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 35 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a45ec0c30a	Merged revisions 174945 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago