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<a href="http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#000">ASTTRUNK-FREEBSD81-I386-266</a>
<span style="font-family: Arial, sans-serif; font-size: 14px;"> has been queued, but there's no agent capable of building it.</span>
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<a href="http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log" style="text-decoration: none; font-family: Arial, sans-serif;text-align:left;font-size:16px;font-weight:bold;color:#036" >Code Changes</a>
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Merged revisions 324479 via svnmerge from <br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.8">https://origsvn.digium.com/svn/asterisk/branches/1.8</a><br/>
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r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line<br/>
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Comments and whitespace in chan_sip.c<br/>
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(324480)
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Merged revisions 324481 via svnmerge from<br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.8">https://origsvn.digium.com/svn/asterisk/branches/1.8</a><br/>
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Also fixed a reference leak in an error path in sip_msg_send().<br/>
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r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines<br/>
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Timout or error on INFO or MESSAGE transaction causes call to be lost.<br/>
<br/>
When exchanging INFO messages within a call, 4xx error causes the call to<br/>
be disconnected although RFC 2976 explicitly states that such transactions<br/>
do not modify the state of the dialog.<br/>
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When exchanging MESSAGE messages within a call, 4xx error causes the call<br/>
to be disconnected. To provide least surprise, we should not disconnect<br/>
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428<br/>
Section 2)<br/>
<br/>
(closes issue ASTERISK-17901)<br/>
Reported by: neutrino88<br/>
<br/>
Review: <a href="https://reviewboard.asterisk.org/r/1257/">https://reviewboard.asterisk.org/r/1257/</a><br/>
Review: <a href="https://reviewboard.asterisk.org/r/1258/">https://reviewboard.asterisk.org/r/1258/</a><br/>
<br/>
JIRA SWP-3486<br/>
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(324482)
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<a href="http://bamboo.asterisk.org/browse/author/twilson" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
twilson</a><br>
Merged revisions 324484 via svnmerge from <br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.8">https://origsvn.digium.com/svn/asterisk/branches/1.8</a><br/>
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines<br/>
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Stop sending IPv6 link-local scope-ids in SIP messages<br/>
<br/>
The idea behind the patch listed below was used, but in a more targeted manner.<br/>
There are now address stringification functions for addresses that are meant to<br/>
be sent to a remote party. Link-local scope-ids only make sense on the machine<br/>
from which they originate and so are stripped in the new functions.<br/>
<br/>
There is also a host sanitization function added to chan_sip which is used<br/>
for when peer and dialog tohost fields or sip_registry hostnames are used to<br/>
craft a SIP message.<br/>
<br/>
Also added are some basic unit tests for netsock2 address parsing.<br/>
<br/>
(closes issue ASTERISK-17711)<br/>
Reported by: ch_djalel<br/>
Patches:<br/>
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)<br/>
<br/>
Review: <a href="https://reviewboard.asterisk.org/r/1278/">https://reviewboard.asterisk.org/r/1278/</a><br/>
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(324487)
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<br/><a href="http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log" style="font-family: Arial, sans-serif; font-size: 13px; color:#036">6 more changes...</a>
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