Merged revisions 55086 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Sat, 17 Feb 2007 01:22:01 +0000 (01:22 +0000)
committerJoshua Colp <jcolp@digium.com>
Sat, 17 Feb 2007 01:22:01 +0000 (01:22 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines

Merged revisions 55073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines

Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index e852a77..f12f81c 100644 (file)
@@ -13093,7 +13093,11 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                        } else if (sipmethod == SIP_NOTIFY) {
                                /* They got the notify, this is the end */
                                if (p->owner) {
-                                       ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
+                                       if (p->refer) {
+                                               if (option_debug)
+                                                       ast_log(LOG_DEBUG, "Got 200 OK on NOTIFY for transfer\n");
+                                       } else
+                                               ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
                                        /* ast_queue_hangup(p->owner); Disabled */
                                } else {
                                        if (!p->subscribed && !p->refer)
@@ -13384,7 +13388,7 @@ static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target
                        ast_log(LOG_DEBUG, "-- No target second channel ---\n");
                ast_log(LOG_DEBUG, "-- END Sip transfer:--------------------\n");
        }
-       if (transferer->chan2) {                        /* We have a bridge on the transferer's channel */
+       if (transferer->chan2 && (ast_bridged_channel(transferer->chan2) == transferer->chan2->_bridge)) { /* We have a bridge on the transferer's channel */
                peera = transferer->chan1;      /* Transferer - PBX -> transferee channel * the one we hangup */
                peerb = target->chan1;          /* Transferer - PBX -> target channel - This will get lost in masq */
                peerc = transferer->chan2;      /* Asterisk to Transferee */