Merged revisions 188947 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines

  Merged revisions 188946 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines

    Fix a bug where a value used to create the channel name was bogus.

    This commit fixes the scenario where an incoming call is authenticated
    using a peer entry. Previously the channel name was created using either
    the username setting from the sip.conf entry or the IP address that the
    call came from. Now the channel name will be created using the peer name
    itself. This commit will not change the way the channel name is generated
    for users or friends.

    (closes issue #14256)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-chname.patch uploaded by Nick (license 657)
    Tested by: Nick_Lewis, file
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188949 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 12578fc2ec968846e021d0a2effbd22dd5036e8e..7a325a17c866ad2555b8b28fb4668196123f8f8c 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -18174,7 +18174,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 
                        make_our_tag(p->tag, sizeof(p->tag));
                        /* First invitation - create the channel */
-                       c = sip_new(p, AST_STATE_DOWN, S_OR(p->username, NULL));
+                       c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL));
                        *recount = 1;
 
                        /* Save Record-Route for any later requests we make on this dialogue */