]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Merged revisions 371358 via svnmerge from
authorAutomerge script <automerge@asterisk.org>
Thu, 16 Aug 2012 19:25:13 +0000 (19:25 +0000)
committerAutomerge script <automerge@asterisk.org>
Thu, 16 Aug 2012 19:25:13 +0000 (19:25 +0000)
file:///srv/subversion/repos/asterisk/branches/10

................
  r371358 | jrose | 2012-08-16 14:05:21 -0500 (Thu, 16 Aug 2012) | 11 lines

  chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header

  Previously the pvt SIP_OUTGOING flag was used instead, which will frequently
  flip during reinvites.

  (closes issue AST-897)
  Reported by: Thomas Arimont
  ........

  Merged revisions 371357 from http://svn.asterisk.org/svn/asterisk/branches/1.8
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371381 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 020833d861bdee2ed1203d684d14dfa9c47c0e17..5521546a437bed8b31811923af8e3635989f1336 100644 (file)
@@ -11395,7 +11395,7 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
                }
                add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
        } else {
-               ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "calling" : "called");
+               ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
 
                switch (lid_pres) {
                case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED: