1705 Commits (3e76d988a54822a9162aa6ac56a32ceff7ea3818)

Author	SHA1	Message	Date
Walter Doekes	ef82f32ca3	Cleanup references to sipusers and sipfriends dynamic realtime families ... Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	38b76521d2	Don't crash on empty notify channel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5f9c4b212d	Fix sequence number overflow over 16 bits causing codec change in RTP packets. ... Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7106cd9f2b	Cleanup reference leaks in res_jabber ... res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ce19768cec	Fix AGI exec Park to honor the Park application parameters. ... The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	91a65d8b3f	Merged revisions 340970 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	d3831204a9	Fixes some support level info so that it can be read by menuselect. ... (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	4e1084a392	Don't skip the query field on a realtime multi query ... There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	63d4530e93	Merged revisions 340108 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	2e35748372	Merged revisions 339505 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ee29743c1f	Only change the capabilities on the gateway when ... the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	9c1362a6d1	Merged revisions 339297 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e71f06475e	Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. ... This function prints a list of caps instead of a hex bitfield. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	36b780871a	Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	0469ffa481	properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	16cf441dcc	Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will ... turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	283f63c594	Remove T38 Gateway capability when detaching framehook. ... SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	34ae017509	Merged revisions 337541 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	42c967eea9	Change strictrtp option to default to yes in the RTP module ... Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5f1a062fa6	Merged revisions 336877 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8285989c1c	Merged revisions 336716 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d287e6116a	Merged revisions 335497 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2ce6c668ca	Merged revisions 334355 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	a5922759ce	Merged revisions 334229 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines Create a local alias for ast_odbc_clear_cache. As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c4cdaf31c6	only alter the gateway_timeout when attching the gateway to a channel ... ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	12e0da4667	Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway). ... Patch by: irroot Review: https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	13288c9c5d	It is possible for the gateway to be attached when the channel is still ... negotiating T.38. This change handles that case. ASTERISK-18329 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6f650edb2b	Merged revisions 333569 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines Accidental use of variable client->status instead of client->state in from ASTERISK-18078 (issue ASTERISK-18078) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	43dee65979	Merged revisions 333378 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines [patch] Buddies are always auto-registered when processing the roster Reporter said autoregister flag was ignored for registering 'buddies' which had a subscription to us. Verified that this was the case and observed how the patch addressed this and made sure it didn't break anything. (closes issue ASTERISK-14233) Reported by: Simon Arlott Patches: asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott Tested by: Jonathan Rose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	77055101cd	Merged revisions 333265 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines Segfault when publishing device states via XMPP and not connected When using publishing device state with res_jabber, Asterisk will attempt to send a device state using the unconnected client using iks_send_raw and crash. This patch checks the validity of the connection before attempting to send the device state. (closes issue ASTERISK-18078) Reported by: Michael L. Young Patches: res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young Tested by: Jonathan Rose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	a0c2e7ac31	Changed the "timeout" option to "gwtimeout". ... ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c35a68f831	Merged revisions 332816 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	f3c6244fa9	add a way to disable and/or modify the gateway timeout ... ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d44e801e6c	Merged revisions 332320 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines Don't read from a disarmed or invalid timerfd Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	6984fbaa95	Merged revisions 331038 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines In-queue MOH stops after a periodic announcement If the seek value is past the end of file when resuming G.722 MOH, MOH will cease to function for the duration of the MOH session through all starts and stops until saved state is cleared. Adjusting the code to guarantee a single valid read (which is already assumed) fixes the bug. (closes issue ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: Jonathan Rose <jrose@digium.com> ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	16cd91b92e	Merged revisions 330648 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug 2011) | 2 lines Convert an error message to actually be helpful. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	d6497d4284	Merged revisions 329991 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file Patch by: tzafrir Reported by: tzafrir (closes issue ASTERISK-18161) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	98c4fee4cb	Merged revisions 328823 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	4c31c3b3a5	Merged revisions 328205 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines Monitor application arguments requirements fixed. Monitor was requiring options in spite of no individual option on Monitor being required. Review: https://reviewboard.asterisk.org/r/1320/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	3f44b08b7b	do v21 detection instead of CED detection for the fax gateway ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b2ad651482	renamed fax_gateway_send_ced() to fax_gateway_request_t38() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c42c024edf	actually do something with the ced timeout, also added more debug output ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4f08a3a8eb	write silence on the channel during t.38 negotiation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	746f93de45	Delay sending an CED tone generated T.38 reinvite to give the CED tone ... generating party time to send its own T.38 reinvite. Also don't forward frames through the gateway if we are negotiating T.38. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	96fad8dba6	fixed wording in a comment ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a86c1d68e9	Moves celt and silk format attribute files into res folder. ... It was inconsistent to have the silk and celt format attribute modules in the format file interpreter folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. ... This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	b5609161e0	Merged revisions 326830 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) | 1 line libgen.h is also needed on Darwin for basename(3) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	c545e3b1c5	Merged revisions 326689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines res_odbc patch by tilghman to fix integers with null values Addresses some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as "(NULL)" rather than an actual NULL. (closes issue #1922STERISK-17791) Reported by: marcelloceschia Patches: 20110505__issue19223.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago