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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines
Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead
of setting the field manually to avoid uninitialized data.
Review: https://reviewboard.asterisk.org/r/1076/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines
Fix for a consistent MulticastRTP channel driver crash due to use of unitilized
data.
(closes issue #18290)
(closes issue #18602)
Reported by: voipgate, wybecom
Review: https://reviewboard.asterisk.org/r/1076/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines
Split _all_ arguments before parsing them.
This fixes multicast RTP paging using linksys mode.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11797)
Reported by: macbrody
Review: https://reviewboard.asterisk.org/r/270/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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