[asterisk-commits] file: branch 1.2 r53103 - /branches/1.2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu Feb  1 16:21:56 2007
New Revision: 53103

URL: http://svn.digium.com/view/asterisk?view=rev&rev=53103
Log:
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.

Modified:
    branches/1.2/channels/chan_sip.c

Modified: branches/1.2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?view=diff&rev=53103&r1=53102&r2=53103
==============================================================================
--- branches/1.2/channels/chan_sip.c (original)
+++ branches/1.2/channels/chan_sip.c Thu Feb  1 16:21:56 2007
@@ -2083,6 +2083,7 @@
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
 		p->jointcapability = p->capability;
+		p->jointnoncodeccapability = p->noncodeccapability;
 		transmit_invite(p, SIP_INVITE, 1, 2);
 		if (p->maxtime) {
 			/* Initialize auto-congest time */