Merged revisions 158053 via svnmerge from
authorMark Michelson <mmichelson@digium.com>
Thu, 20 Nov 2008 17:39:06 +0000 (17:39 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 20 Nov 2008 17:39:06 +0000 (17:39 +0000)
commit7a554a7386478fe93db31dd2ba5a421063f6cdbb
tree1025fd910d77eff81d9815d9107328a46b4ec689
parentd12263a16a2eb4e3877a12a63b5531395d34ad4b
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue #13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
apps/app_dial.c
channels/chan_sip.c