Joshua Colp
e85296e244
Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-19 15:41:24 +00:00
Mark Michelson
dce6a54a4a
Trunk implementation of setting an alternate RTP source.
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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
2009-06-18 15:20:17 +00:00
Joshua Colp
4900d37d71
Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
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(closes issue #15243 )
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-02 13:12:59 +00:00
Russell Bryant
f6bdacebe4
Don't do a pointer comparison before setting the remote address.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 22:42:15 +00:00
Joshua Colp
c1073e0ac3
Merged revisions 195095 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines
Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited.
(closes issue #13569 )
Reported by: bkw918
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 13:56:16 +00:00
Russell Bryant
2c1ffef923
Resolve Solaris build issues and add some API documentation.
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(issue #14981 )
Reported by: snuffy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29 08:51:21 +00:00
Joshua Colp
aaf1566222
Change how we set the local and remote address.
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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
2009-04-10 18:14:47 +00:00
Joshua Colp
8e4b5df187
Fix some uninitialized memory notices that appeared under valgrind.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 18:02:44 +00:00
Joshua Colp
b0637fe624
Fix a bug where we would native bridge when we did not want to.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 18:12:28 +00:00
Joshua Colp
0ab599bf94
Turn a warning message into a debug message and do not treat two situations as errors when they are not.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 16:27:36 +00:00
Joshua Colp
9597e33a63
Pass the correct value to sizeof when copying address information.
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(issue #14827 )
Reported by: pj
Patches:
14827.diff uploaded by file (license 11)
Tested by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-06 13:23:12 +00:00
Joshua Colp
63de834395
Merge in the RTP engine API.
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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
2009-04-02 17:20:52 +00:00
