[test-results] [Bamboo] Asterisk - Trunk > FreeBSD 8.1 > #405 has FAILED (62 tests failed, 1 failures were new). Change made by 4 authors.

[Bamboo]

-----------------------------------------------------------------------
Asterisk - Trunk > FreeBSD 8.1 > #405 failed.
-----------------------------------------------------------------------
This build occurred because it is a dependant of ASTTRUNK-LUCID-1299.
62/178 tests failed, 1 failure was new.

http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-405/


--------------
Failing Jobs
--------------
  - i386 (Default Stage): 62 of 178 tests failed.


--------------
Code Changes
--------------
rmudgett (348953):

>Fix extension state callback references in chan_sip.
>
>Chan_sip gives a dialog reference to the extension state callback and
>assumes that when ast_extension_state_del() returns, the callback cannot
>happen anymore.  Chan_sip then reduces the dialog reference count
>associated with the callback.  Recent changes (ASTERISK-17760) have
>resulted in the potential for the callback to happen after
>ast_extension_state_del() has returned.  For chan_sip, this could be very
>bad because the dialog pointer could have already been destroyed.
>
>* Added ast_extension_state_add_destroy() so chan_sip can account for the
>sip_pvt reference given to the extension state callback when the extension
>state callback is deleted.
>
>* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
>and handle_statechange() now that the struct ast_state_cb has a destructor
>to call.
>
>* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
>for a successful registration.
>
>* Fixed pbx.c statecbs_cmp() to compare the correct information.  The
>passed in value to compare is a change_cb function pointer not an object
>pointer.
>
>* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
>AST_EXTENSION_REMOVED with locks held.  Chan_sip is notorious for
>deadlocking when those locks are held during the callback.
>
>* Removed unused lock declaration for the pbx.c store_hints list.
>
>(closes issue ASTERISK-18844)
>Reported by: rmudgett
>
>Review: https://reviewboard.asterisk.org/r/1635/
>........
>
>Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 348952 from http://svn.asterisk.org/svn/asterisk/branches/10
>

wedhorn (348849):

>Fix segfault on answer.
>
>Only update/change RTP source if RTP has already been started and 
>connected to the subchannel.
>
>

Matthew Jordan (348890):

>Fix for memory leaks / cleanup in cel_pgsql
>
>There were a number of issues in cel_pgsql's pgsql_log method:
>* If either sql or sql2 could not be allocated, the method would return while
>the pgsql_lock was still locked
>* If the execution of the log statement succeeded, the sql and sql2 structs
>were never free'd
>* Reconnection successes were logged as ERRORs.  In general, the severity of
>several logging statements was reduced
>
>(closes issue ASTERISK-18879)
>Reported by: Niolas Bouliane
>Tested by: Matt Jordan
>
>Review: https://reviewboard.asterisk.org/r/1624/
>........
>
>Merged revisions 348888 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 348889 from http://svn.asterisk.org/svn/asterisk/branches/10
>


--------------
Tests
--------------
New Test Failures (1)
   - AsteriskTestSuite: S/apps/confbridge/confbridge nominal
Existing Test Failures (61)
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail priority
   - AsteriskTestSuite: S/apps/voicemail/check voicemail reply
   - AsteriskTestSuite: S/fastagi/connect
   - AsteriskTestSuite: S/apps/voicemail/check voicemail new user
   - AsteriskTestSuite: S/apps/voicemail/func vmcount
   - AsteriskTestSuite: S/fastagi/stream-file
   - AsteriskTestSuite: S/directed pickup
   - AsteriskTestSuite: S/apps/directory operator exit
   - AsteriskTestSuite: S/channels/ s i p/options
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding
   - AsteriskTestSuite: S/apps/voicemail/check voicemail dialout
   - AsteriskTestSuite: S/channels/ s i p/nat supertest
   - AsteriskTestSuite: S/apps/incomplete/sip incomplete
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding auto urgent
   - AsteriskTestSuite: S/apps/voicemail/check voicemail envelope
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options change password
   - AsteriskTestSuite: S/channels/ s i p/sip register domain acl
   - AsteriskTestSuite: S/fax/gateway t38 g711
   - AsteriskTestSuite: S/fax/gateway g711 t38
   - AsteriskTestSuite: S/channels/ s i p/realtime nosipregs
   - AsteriskTestSuite: S/fax/gateway native t38 ced
   - AsteriskTestSuite: S/udptl
   - AsteriskTestSuite: S/apps/directory context operator exit
   - AsteriskTestSuite: S/fax/gateway timeout5

--
This message is automatically generated by Atlassian Bamboo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/test-results/attachments/20111227/6b1227f6/attachment-0001.htm>