[asterisk-commits] dvossel: trunk r283596 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Wed Aug 25 17:59:15 2010
New Revision: 283596

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283596
Log:
Merged revisions 283595 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines
  
  Merged revisions 283594 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines
    
    Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
    
    When pedantic mode is used, the dialog-info xml generated during a
    ringing event must contain the to and from tag values.  Otherwise if
    a pickup occurs using INVITE with replaces, Astrisk will not be able
    to locate the subscription.
  ........
................

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=283596&r1=283595&r2=283596
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Aug 25 17:59:15 2010
@@ -11688,8 +11688,14 @@
 
 				/* We create a fake call-id which the phone will send back in an INVITE
 				   Replaces header which we can grab and do some magic with. */
+				if (sip_cfg.pedanticsipchecking) {
+					ast_str_append(tmp, 0, "<dialog id=\"%s\" call-id=\"pickup-%s\" local-tag=\"%s\" remote-tag=\"%s\" direction=\"recipient\">\n",
+						exten, p->callid, p->theirtag, p->tag);
+				} else {
+					ast_str_append(tmp, 0, "<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n",
+						exten, p->callid);
+				}
 				ast_str_append(tmp, 0,
-						"<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n"
 						"<remote>\n"
 						/* See the limitations of this above.  Luckily the phone seems to still be
 						   happy when these values are not correct. */
@@ -11700,7 +11706,7 @@
 						"<identity>%s</identity>\n"
 						"<target uri=\"%s\"/>\n"
 						"</local>\n",
-						exten, p->callid, local_display, local_target, local_target, mto, mto);
+						local_display, local_target, local_target, mto, mto);
 			} else {
 				ast_str_append(tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", exten);
 			}
@@ -20983,7 +20989,7 @@
 			struct sip_pvt *subscription = NULL;
 			replace_id += 7; /* Worst case we are looking at \0 */
 
-			if ((subscription = get_sip_pvt_byid_locked(replace_id, NULL, NULL)) == NULL) {
+			if ((subscription = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
 				ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
 				transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
 				error = 1;