[asterisk-commits] file: trunk r91440 - in /trunk: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Dec 6 10:18:49 CST 2007


Author: file
Date: Thu Dec  6 10:18:49 2007
New Revision: 91440

URL: http://svn.digium.com/view/asterisk?view=rev&rev=91440
Log:
Merged revisions 91439 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines

Add support for accepting and sending T.38 in the initial INVITE.
(closes issue #9402)
Reported by: thdei

........

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

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=91440&r1=91439&r2=91440
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Dec  6 10:18:49 2007
@@ -4722,6 +4722,10 @@
 		pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
 	if (i->rtp)
 		ast_jb_configure(tmp, &global_jbconf);
+
+	/* If the INVITE contains T.38 SDP information set the proper channel variable so a created outgoing call will also have T.38 */
+	if (i->udptl && i->t38.state == T38_PEER_DIRECT)
+		pbx_builtin_setvar_helper(tmp, "_T38CALL", "1");
 
 	/* Set channel variables for this call from configuration */
 	for (v = i->chanvars ; v ; v = v->next)
@@ -13613,6 +13617,20 @@
 			if (p->owner && !req->ignore)
 				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 			p->needdestroy = 1;
+		} else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
+			/* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
+			   right now we can't fall back to audio so totally abort.
+			*/
+			p->t38.state = T38_DISABLED;
+			/* Try to reset RTP timers */
+			ast_rtp_set_rtptimers_onhold(p->rtp);
+			ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n");
+
+			/* The dialog is now terminated */
+			if (p->owner && !req->ignore)
+				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+			p->needdestroy = 1;
+			sip_alreadygone(p);
 		} else {
 			/* We can't set up this call, so give up */
 			if (p->owner && !req->ignore)




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