mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
193 Commits (bcfe1ecaed2bf6441fcee2bc4e1edb97ac86e631)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kevin P. Fleming
|
2a877c8fcb |
Merged revisions 180373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
|
239ad71be7 |
Merged revisions 178374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
07b9f97f48 |
Merged revisions 178142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 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 |
18 years ago |
Joshua Colp
|
ebdd169c19 |
Merged revisions 170240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | 14 lines Merged revisions 170239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines Don't crash if RTCP is not enabled on an RTP structure but statistics are output. (closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
7333579bf9 |
Merged revisions 165599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines Merged revisions 165591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@165603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
db16f8c6bc |
Merged revisions 162656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change. (closes issue #12983) Reported by: vt Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
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3948ee149b |
Merged revisions 162205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment. (closes issue #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file (license 11) Tested by: ip-rob, bujones ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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a38d0e6fa1 |
Merged revisions 162197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
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704a528b72 |
Merged revisions 161014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines Merged revisions 161013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines (closes issue #13835) Reported by: matt_b Tested by: jpeeler This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure. Closes AST-142. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@161015 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
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c8ebf2045a |
Merged revisions 154060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@154062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Steve Murphy
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062d3041fe |
Merged revisions 147807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Sean Bright
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e509e5b79f |
Merged revisions 138476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Mark Michelson
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b9dfa2fe6c |
Merged revisions 136063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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f68d786c86 |
Merged revisions 129437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines Merged revisions 129436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@129438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Brett Bryant
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b065565bd4 |
Merged revisions 129045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines Janitor project to convert sizeof to ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@129046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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07126be389 |
Merged revisions 125277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) | 15 lines Merged revisions 125276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines Check for rtcp structure before trying to delete schedule. (closes issue #12872) Reported by: destiny6628 Patches: 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) Tested by: destiny6628 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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8b3150e282 |
Merged revisions 116469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines 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/branches/1.6.0@116470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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8d1e1eb09e |
Merged revisions 114101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines 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/branches/1.6.0@114102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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f850d7d9c0 |
Merged revisions 114024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix spelling of existent in a few places. (closes issue #12409) Reported by: candlerb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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8566c2eb9e |
Merged revisions 112210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | 12 lines 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/branches/1.6.0@112211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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aaf6b67e5c |
Merged revisions 110020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | 14 lines 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/branches/1.6.0@110021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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627d3f04d4 |
Merged revisions 109390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines 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/branches/1.6.0@109392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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d334a88e36 |
Merged revisions 106607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008) | 11 lines 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/branches/1.6.0@106608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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ccf90e8186 |
Merged revisions 106501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 | russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines 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/branches/1.6.0@106502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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90046983f7 |
Merged revisions 106239 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines 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/branches/1.6.0@106318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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78393c6698 |
Merged revisions 105933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines 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/branches/1.6.0@106310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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9c09333526 |
Merged revisions 105840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 | tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines Whitespace changes only ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@106306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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83942c273a |
Merged revisions 105677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) | 10 lines 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/branches/1.6.0@106300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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b11caafd7e |
Merged revisions 105675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines 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/branches/1.6.0@106299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Tilghman Lesher
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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 |
19 years ago |
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Joshua Colp
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c81350d6f6 |
Just some minor coding style cleanup...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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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 |
19 years ago |
Olle Johansson
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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 |
19 years ago |
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Olle Johansson
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e7bcc4e96c |
Formatting fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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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 |
19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Russell Bryant
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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 |
19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Olle Johansson
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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
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19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Tilghman Lesher
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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 |
19 years ago |
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Joshua Colp
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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 |
19 years ago |
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Olle Johansson
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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 |
19 years ago |
Luigi Rizzo
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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 |
19 years ago |
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Luigi Rizzo
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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 |
19 years ago |
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Luigi Rizzo
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5663ff6518 |
fix breakage induced by previous mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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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 |
19 years ago |