is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to
transmit_reinvite_with_sdp
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lock when needed - when we remove the dialog from the dialog list
If this doesn't lead to severe problems, it might help with some locking issues
in 1.4/1.2.
- Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no
one understands it... ;-)
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see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216, jmls reported and submitted)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines
We should always be using _exit() after a fork() or vfork() instead of exit().
This is because exit() does some extra cleanup which in some implementations
of vfork(), for example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
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application by configuring them in voicemail.conf (issue #7415, patch by
fkasumovic, with some fixes and documentation updates by myself)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines
Instead of iterating all of the options once to look for jitterbuffer options,
and then again for everything else, move the processing of jitterbuffer
options into the main loop so that there are no erroneous messages about
ignoring unknown options. (issue #8226)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | 11 lines
- If the source has no audio or no video portion, do not call powerof() to
get the format index.
- Don't run through the audio and video loops if there is no audio or video
portion of the source
If 0 is passed to powerof, it will return -1. This value of -1 was then being
used as an array index in these loops, which caused a crash on some systems.
Other than this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is not offered in
the list of acceptable destination formats.
(fixes issue #8231)
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