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hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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destroy it.
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need to be retransmitted.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line
This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels).
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used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
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Will check if I can solve this with less changes in 1.2.
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- Move handling of REFER response to handle_response_refer()
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- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
something that video phones support in the RTP stream.
I now this is a big architectual change at this stage for 1.4, but decided it was needed
to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample
Issue 7679 in the bug tracker. Please test.
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with queues and SIP device states
- Remove support for T.38 early media, since it's impossible.
(Two patches in one - extra friday evening offer due to being off line from svn today... :-)
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(issue #8448 reported by phsultan)
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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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separate line
in the REGISTER request. Imported from 1.2.
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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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subscription support totally or per peer in sip.conf with allowsubscribe = yes | no
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line
fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm
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treat that as
early media. (discovered at the AVTF meeting in Paris).
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destruction (rizzo)
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(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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on name. (issue #8228 reported by dean bath)
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database lookups
for all those realtime users out there.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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- Update docs on canreinvite. "nonat" is the recommended setting for most users with
phones behind a NAT.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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- Never send reply to ACK (again...)
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limitation of file descriptors.
(imported from 1.2)
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anyway, comparing it to 'signed long' is not useful)
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2. A video codec is enabled on the dialog
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OPTION packet from peerpoke
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines
Having more than 255 old messages caused corruption in the new/old count
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- Declare errmsg as int. Oops.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) | 2 lines
Don't play dialtone if the seizing the channel fails (Bug 7754)
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Final response to a re-invite does not mean that the session dies, only that
the re-invite fails.
- Keep RTP active during processing of T.38 re-invite. If the re-invite fails,
RTP needs to remain as before the re-invite.
Issue 8338 - darren1713. Please test.
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-Add some comments to t.38 code
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