[asterisk-commits] dvossel: trunk r282237 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Fri Aug 13 13:58:49 2010
New Revision: 282237

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=282237
Log:
Merged revisions 282236 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
  
  Merged revisions 282235 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
    
    only do magic pickup when notifycid is enabled
    
    A new way of doing BLF pickup was introduced into 1.6.2.  This feature
    adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
    a subscriber that a device is ringing.  This option should only be enabled
    when the new 'notifycid' option is set... but this was not the case.  Instead
    the call-id value was included for every RINGING Notify message, which
    caused a regression for people who used other methods for call pickup.
    
    (closes issue #17633)
    Reported by: urosh
    Patches:
          chan_sip.txt uploaded by urosh (license )
          blf_cid_issue.diff uploaded by dvossel (license 671)
    Tested by: dvossel, urosh, okrief, alecdavis
  ........
................

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=282237&r1=282236&r2=282237
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Aug 13 13:58:49 2010
@@ -11663,23 +11663,26 @@
 					ast_channel_unlock(caller);
 					caller = ast_channel_unref(caller);
 				}
-			}
-
-			/* 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. */
-			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. */
-					"<identity display=\"%s\">%s</identity>\n"
-					"<target uri=\"%s\"/>\n"
-					"</remote>\n"
-					"<local>\n"
-					"<identity>%s</identity>\n"
-					"<target uri=\"%s\"/>\n"
-					"</local>\n",
-					exten, p->callid, local_display, local_target, local_target, mto, mto);
+
+				/* 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. */
+				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. */
+						"<identity display=\"%s\">%s</identity>\n"
+						"<target uri=\"%s\"/>\n"
+						"</remote>\n"
+						"<local>\n"
+						"<identity>%s</identity>\n"
+						"<target uri=\"%s\"/>\n"
+						"</local>\n",
+						exten, p->callid, local_display, local_target, local_target, mto, mto);
+			} else {
+				ast_str_append(tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", exten);
+			}
+
 		} else {
 			ast_str_append(tmp, 0, "<dialog id=\"%s\">", exten);
 		}