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Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
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don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created
(closes issue ASTERISK-23460)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-23460-2.patch
Tested by: Dmitry Melekhov
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Merged revisions 411531 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 411532 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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options are documented in config sample
sample config rename to proper name - ooh323.conf
To change media address ooh323 send empty TCS if there was
completed TCS exchange or send facility forwardedelements
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.
If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange
Review: https://reviewboard.asterisk.org/r/1607/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/10
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r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, 05 Aug 2011) | 11 lines
Merged revisions 330827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines
change gk client behaivour on rrq/grq failures to setup timers
and next tries after timeout instead of complete failure in the ooh323
stack
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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change malloc to ast_calloc again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Now that these files are in the tree, they should prefer the tree's local
copy of all Asterisk headers over any that may be installed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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incorrect q.931 message order filtered on incoming calls (first msg must be setup,
next must be not setup)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Many architectural and functional changes.
Main changes are threading model chanes (many thread in ooh323 stack
instead of one), modifications and improvements in signalling part,
additional codecs support (726, speex), t38 mode support.
This module tested and used in production environment.
(closes issue #15285)
Reported by: may213
Tested by: sles, c0w, OrNix
Review: https://reviewboard.asterisk.org/r/324/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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