[test-results] [Bamboo] No agents to build plan Asterisk - Trunk - FreeBSD 8.1 - i386

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ASTTRUNK-FREEBSD81-I386-266 has been queued, but there's no agent capable of building it.
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http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log

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Code Changes
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rmudgett (324480):

>Merged revisions 324479 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/1.8
>
>........
>  r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line
>  
>  Comments and whitespace in chan_sip.c
>........
>

rmudgett (324482):

>Merged revisions 324481 via svnmerge from
>https://origsvn.digium.com/svn/asterisk/branches/1.8
>
>Also fixed a reference leak in an error path in sip_msg_send().
>
>........
>  r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
>
>  Timout or error on INFO or MESSAGE transaction causes call to be lost.
>
>  When exchanging INFO messages within a call, 4xx error causes the call to
>  be disconnected although RFC 2976 explicitly states that such transactions
>  do not modify the state of the dialog.
>
>  When exchanging MESSAGE messages within a call, 4xx error causes the call
>  to be disconnected.  To provide least surprise, we should not disconnect
>  the call since a MESSAGE is like INFO in this case.  (Implied by RFC 3428
>  Section 2)
>
>  (closes issue ASTERISK-17901)
>  Reported by: neutrino88
>
>  Review: https://reviewboard.asterisk.org/r/1257/
>  Review: https://reviewboard.asterisk.org/r/1258/
>
>  JIRA SWP-3486
>........
>

twilson (324487):

>Merged revisions 324484 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/1.8
>
>........
>  r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
>  
>  Stop sending IPv6 link-local scope-ids in SIP messages
>  
>  The idea behind the patch listed below was used, but in a more targeted manner.
>  There are now address stringification functions for addresses that are meant to
>  be sent to a remote party. Link-local scope-ids only make sense on the machine
>  from which they originate and so are stripped in the new functions.
>  
>  There is also a host sanitization function added to chan_sip which is used
>  for when peer and dialog tohost fields or sip_registry hostnames are used to
>  craft a SIP message.
>  
>  Also added are some basic unit tests for netsock2 address parsing.
>  
>  (closes issue ASTERISK-17711)
>  Reported by: ch_djalel
>  Patches:
>        asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
>  
>  Review: https://reviewboard.asterisk.org/r/1278/
>........
>


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