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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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(imported from 1.4)
- Small fixes to limitonpeer
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines
Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)
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Luigi, but hit me in the back today.
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source, not the configuration option name so let's turn it back to srvlookup instead of global_srvlookup. (issue #8442 reported by jtodd)
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that occurs for every incoming media frame.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 lines
fix bug 7450 - Parsing fails if From header contains angle brackets
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
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doing busy-wait on the output audio device.
As it is set now, it tries to push a frame every 10ms,
which is still too frequent but avoids deep restructuring
of the code (which i should do, though).
Note, this is only for ring tones, regular audio coming
from the network is still delivered as soon as it is
available.
Eventually this could well end up in the 1.4 branch, but
since i am probably the only user of chan_oss there isn't
much urgency to do that.
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if we're bringing the
link up for the first time.
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BRI). (#8077) Thanks Tzafrir.
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structure directly.
Also move the default checking to where it gets set initially.
Fixes suggested by file.
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Issue #8369, patch by ssuehring, modified slightly by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2 lines
Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath)
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for realtime users.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, 16 Nov 2006) | 10 lines
Merged revisions 47750 via svnmerge from
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r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines
Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell)
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1.2. This is a bit more
clean platform for the handling of BYE/CANCEL than what we had. It might also need to changes
in other parts of the code, since we know the state of the INVITE transaction.
Please observe that this is is not dialog states at all, this is INVITE transaction states.
Hello Michael Proctor, and thank you! :-)
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- Add docs on canreinvite
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, 15 Nov 2006) | 10 lines
Merged revisions 47711 via svnmerge from
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r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines
Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me)
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- Improve SIP history for debugging
(Imported from 1.4)
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of file handles is one cause)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14 Nov 2006) | 2 lines
silence compiler warning on 64-bit platforms (this variable is an 'int' anyway, comparing it to 'signed long' is not useful)
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r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2 lines
Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog
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overwrite the initreq...
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