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Some PBXes require this for call status checking
(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
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Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 344159 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/10
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r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 333947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed
before ast_cond_wait called to wait this
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r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines
security fix. really drop call if signalling addr is not same as socket
addr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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small fixes.
Interpret remote side H.225 version.
Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.
Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.
(closes issue #18542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
(closes issue #18433)
Reported by: MrHanMan
Patches:
tcs-msd-h245-3.patch uploaded by may213 (license 454)
Tested by: MrHanMan, may213
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
Merged revisions 284593,284595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265525 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242933 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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when we decode received q931 packet we must do callbacks and
when we print sended q931 packet we must not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239037 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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