Merged revisions 378565,378585 via svnmerge from

file:///srv/subversion/repos/asterisk/trunk

................
  r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines
  
  Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
  
  On a multihomed server when sending a NOTIFY message, we were not figuring out
  which network should be used to contact the peer.
  
  This patch fixes the problem by calling ast_sip_ouraddrfor() and then
  build_via() so that our NOTIFY message contains the correct IP address.
  
  Also, a debug message is being added to help follow the call-id changes that
  occur.  This was helpful for confirming that the IP address was set properly
  since the call-id contains the IP address.  It also will be helpful for
  troubleshooting purposes when following a call in the debug logs.
  
  (closes issue ASTERISK-20805)
  Reported by: Bryan Hunt
  Tested by: Bryan Hunt, Michael L. Young
  Patches:
      asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2255/
  ........
  
  Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11
................
  r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines
  
  Fix pjproject compilation in certain circumstances
  
  On a fresh checkout of Asterisk 11, running make before ./configure
  could cause the pjproject subdirectory to get in an odd state that
  would prevent compilation. This patch by Tilghman prevents that from
  occurring.
  
  (closes issue ASTERISK-20681)
  Patch-by: Tilghman Lesher
  ........
  
  Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 11 insertions, 1 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 18c14eacd..528403d8a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8553,6 +8553,7 @@ static void change_callid_pvt(struct sip_pvt *pvt, const char *callid)
 {
 	int in_dialog_container;
 	int in_rtp_container;
+	char *oldid = ast_strdupa(pvt->callid);
 
 	ao2_lock(dialogs);
 	ao2_lock(dialogs_rtpcheck);
@@ -8573,6 +8574,10 @@ static void change_callid_pvt(struct sip_pvt *pvt, const char *callid)
 	}
 	ao2_unlock(dialogs_rtpcheck);
 	ao2_unlock(dialogs);
+
+	if (strcmp(oldid, pvt->callid)) {
+		ast_debug(1, "SIP call-id changed from '%s' to '%s'\n", oldid, pvt->callid);
+	}
 }
 
 /*! \brief Build SIP Call-ID value for a REGISTER transaction */
@@ -21553,7 +21558,12 @@ static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
 			}
 		}
 
-		/* Recalculate our side, and recalculate Call ID */
+		/* Now that we have the peer's address, set our ip and change callid */
+		ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
+		build_via(p);
+
+		change_callid_pvt(p, NULL);
+
 		ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
 		sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
 		transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);