Merged revisions 279227 via svnmerge from
authorRichard Mudgett <rmudgett@digium.com>
Fri, 23 Jul 2010 22:24:52 +0000 (22:24 +0000)
committerRichard Mudgett <rmudgett@digium.com>
Fri, 23 Jul 2010 22:24:52 +0000 (22:24 +0000)
commitff2dc29d882e12817995ffeec8cd124151415bfa
tree52d8f485c9ea5616f19a0bd633b7650059cf99a7
parentd7ca69ceeac04c8a350c33e59c951a08affaaf1f
Merged revisions 279227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines

  Merged revisions 279207 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines

    Merged revisions 279206 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines

      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/trunk@279245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
apps/app_dial.c
apps/app_queue.c