[asterisk-commits] dvossel: branch 1.8 r282236 - in /branches/1.8: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Fri Aug 13 13:58:10 2010
New Revision: 282236

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=282236
Log:
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:
    branches/1.8/   (props changed)
    branches/1.8/channels/chan_sip.c

Propchange: branches/1.8/
------------------------------------------------------------------------------
--- branch-1.6.2-merged (original)
+++ branch-1.6.2-merged Fri Aug 13 13:58:10 2010
@@ -1,1 +1,1 @@
-/branches/1.6.2:1-279056,279207,279501,279561,279597,279609,279657,279784,279849,279946,280089,280160,280193,280229,280231,280306,280345,280449,280551,280671,280739,280983,281051,281391,281430,281567,281574,281722,281763,281873,281912,282130
+/branches/1.6.2:1-279056,279207,279501,279561,279597,279609,279657,279784,279849,279946,280089,280160,280193,280229,280231,280306,280345,280449,280551,280671,280739,280983,281051,281391,281430,281567,281574,281722,281763,281873,281912,282130,282235

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=282236&r1=282235&r2=282236
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Fri Aug 13 13:58:10 2010
@@ -11660,23 +11660,26 @@
 					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);
 		}