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(parent: 7bc66e1) | patch
SIP promiscuous redirect could fail to dial the redirect.
authorRichard Mudgett <rmudgett@digium.com>
Fri, 23 Jul 2010 21:56:44 +0000 (21:56 +0000)
committerRichard Mudgett <rmudgett@digium.com>
Fri, 23 Jul 2010 21:56:44 +0000 (21:56 +0000)
commit08883af231fe9b12691b992dea1adb4ad513c19b
tree2956574fba525068198623d51251445d4f56b2eftree
parent7bc66e1d7751d2cfb8d9a57a664c66c4b34a4e72commit | diff
SIP promiscuous redirect could fail to dial the redirect.

The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable.  The two variables are not equivalent if the call_forward string
included a channel technology specifier.  e.g., SIP/200

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
apps/app_dial.c diff | blob | blame | history
apps/app_queue.c diff | blob | blame | history
Mirror of https://github.com/asterisk/asterisk.git