asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	f59fe83c56	More 32->64 bit codec conversions. In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	cf87d81e9d	Merged revisions 231441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231441 \| dvossel \| 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) \| 11 lines fixes crash caused by RTP comfort noise payload greater than 24 bytes AST-2009-010 (closes issue #16242) Reported by: amorsen Patches: issue16242.diff uploaded by oej (license 306) Tested by: amorsen, oej, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	844a01b27e	Add an "Asterisk Architecture Overview" section to the doxygen documentation. This is a side project I've been poking at this week. The intent is to discuss Asterisk architecture in a top down fashion to help new developers understand how Asterisk is put together. There is a ton of stuff to write about, so this will just continue to evolve over time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	092a118d89	Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 \| kpfleming \| 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) \| 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	717d2ec3c9	Remove spurious debug git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	10ce6cd757	Use rtp properties instead of adding a callback Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	865daf4858	Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 \| twilson \| 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) \| 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	3c04a79abf	use the actual given ip address for 'rtp set debug ip <foo>' instead of the word 'ip' (closes issue #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ed8ccbdb73	Gracefully handle malformed RTP text packets. AST-2009-004 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba2a8457b8	Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 \| dvossel \| 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) \| 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	1179ecf165	Merged revisions 194208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 \| file \| 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) \| 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	973b36a3c7	Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	aaf1566222	Change how we set the local and remote address. The code will now only change the address and port. It will not overwrite any other values. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	8e4b5df187	Fix some uninitialized memory notices that appeared under valgrind. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	0ab599bf94	Turn a warning message into a debug message and do not treat two situations as errors when they are not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	c02b56f7bc	Fix a log message getting output when it should not have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

21 Commits (f59fe83c56f6539c09eb068a94d2db60bfb18f17)