[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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