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authorMark Michelson <mmichelson@digium.com>2008-11-20 17:39:06 +0000
committerMark Michelson <mmichelson@digium.com>2008-11-20 17:39:06 +0000
commit7a554a7386478fe93db31dd2ba5a421063f6cdbb (patch)
tree1025fd910d77eff81d9815d9107328a46b4ec689 /cdr/cdr_csv.c
parentd12263a16a2eb4e3877a12a63b5531395d34ad4b (diff)
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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