179 Commits (d32c2d6fd9d6970ad2cde1b29fefff445dc98fb1)

Author	SHA1	Message	Date
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause ... - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	08f91c1192	Merged revisions 116463 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	bb386c84e7	Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream ... Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5fff9c7304	Merged revisions 114100 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this. (closes issue #12353) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4a21c5dd22	Fix spelling of existent in a few places. ... (closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0d7cfae6b6	Merged revisions 112209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ef4eff9a9b	Add the "config reload <conffile>" command, which allows you to tell Asterisk ... to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3e439e9616	Merged revisions 110019 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	10cdbe28a8	Merged revisions 109386 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a60f591c72	Merged revisions 106606 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines Properly initialize rtp->schedid (Closes issue #12154) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5ca5d97673	Merge changes from team/russell/g722-sillyness ... ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	496adc6fc0	Merged revisions 106235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a760a033e9	Merged revisions 105932 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	cfc1df4c1a	Whitespace changes only ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3b070a815d	Merged revisions 105676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4de0d8368f	Merged revisions 105674 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	dca12f4aa7	Fix T38 passthrough regression introduced by state changes. ... (closes issue #12078) Reported by: dimas Patches: v1-12078.patch uploaded by dimas (license 88) (closes issue #12074) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2c3c489ade	Merged revisions 103780 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines When a SIP channel is being auto-destroyed, it's possible for it to still be in bridge code. When that happens, we crash. Delay the RTP destruction until the bridge is ended. (closes issue #11960) Reported by: norman Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) Tested by: norman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c81350d6f6	Just some minor coding style cleanup... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1ec8cb41a8	Merge changes from team/mvanbaak/cli-command-audit ... (closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	94325433a2	- doxygen fixes ... - change function to void because it always returned the same value and no one read it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	e7bcc4e96c	Formatting fixes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	ac699196f5	Merged revisions 100465 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines When deleting a task from the scheduler, ignoring the return value could possibly cause memory to be accessed after it is freed, which causes all sorts of random memory corruption. Instead, if a deletion fails, wait a bit and try again (noting that another thread could change our taskid value). (closes issue #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, stuarth` ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3bf7daa0c0	Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which drops packets that do not come from the remote party. ... (closes issue #8952) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	2ee416a55a	Merged revisions 98958 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 lines Add two more SDP names for ulaw and alaw. (closes issue #11777) Reported by: tootai ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	4fb04cb58a	Merged revisions 98943 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	8e0dbcf7d7	Merged revisions 98325 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	17afebc1a6	HUGE improvements to QoS/CoS handling by IgorG ... - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	8a7064d3fc	Merged revisions 92204 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much. (closes issue #11483) Reported by: revolution Patches: rtp.diff uploaded by revolution (license 346) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	77ec19e255	Merged revisions 91637 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) | 5 lines At the end of a call, when we're reporting, RTCP may already be partially torn down, so check for NULL dereference Reported by: blitzrage Patch by: tilghman (Closes issue #11450) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	985c9f5cfe	Merged revisions 90588 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	df7ba90b20	The following patch with updates for trunk. Works much better in trunk. ... Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	e0ff5fef5c	remove a bunch of useless #include "options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	9335ace850	another bunch of include removals (errno.h and asterisk/logger.h) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	5663ff6518	fix breakage induced by previous mistake ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	5490960453	remove a bunch of duplicate includes ... Reproduce with grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7c56918262	Commit some cleanups to the format type code. ... - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	255e26c480	Drop the RTCP Read too short message to debug. There are some phones out there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ebe4050128	Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b0f3e6097e	Convert NEW_CLI to AST_CLI. ... Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	65761cbd7a	More changes to NEW_CLI. ... Also fixes a few cli messages and some minor formatting. (closes issue #11001) Reported by: seanbright Patches: newcli.1.patch uploaded by seanbright (license 71) newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch uploaded by seanbright (license 71) newcli.5.patch uploaded by seanbright (license 71) newcli.6.patch uploaded by seanbright (license 71) newcli.7.patch uploaded by seanbright (license 71) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	a5122f03ad	Merged revisions 85559 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 lines Bring both DTMF begin and end frames up through to the core for DTMF feature handling. (closes issue #10826) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c7ea8f9c87	Merged revisions 85552 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 lines If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work. (closes issue #10943) Reported by: julianjm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6f2e7b4310	Merged revisions 85057 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4 lines Only update codec information if the channel has a technology private structure. (issue #10915) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	de64c85b54	Merged revisions 85023 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4 lines Update codec information as well as address when doing hold reinvites. (issue #10868) Reported by: mavince ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	5b3347c715	Merged revisions 84818 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 lines Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite. (closes issue #10868) Reported by: mavince ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	cfc8e90501	Merged revisions 84581 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007) | 2 lines When an RFC 2833 event is sent that we don't recognize, ignore it, don't queue a NULL digit (closes issue #10877) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	fe1d4b1d04	Don't swap channel priority if using epoll as polling should/will only happen off the first channel. ... (closes issue #10867) Reported by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d78463be1e	Corydon posted this janitor project to the bug tracker and mvanbaak provided ... a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string (closes issue #10843) Reported by: Corydon76 Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago