From: David Vossel <dvossel@digium.com>
Date: Fri, 13 Aug 2010 18:58:10 +0000 (+0000)
Subject: Merged revisions 282235 via svnmerge from
X-Git-Tag: 1.8.0-beta4~36
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=fbfafb59bab28fa5f851f938bfbc3e973ebf4826;p=thirdparty%2Fasterisk.git

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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c0d0924702..7fe6a635ae 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11660,23 +11660,26 @@ static void state_notify_build_xml(int state, int full, const char *exten, const
 					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);
+			} else {
+				ast_str_append(tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", exten);
 			}
 
-			/* 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\">", exten);
 		}