[asterisk-commits] trunk r36171 - /trunk/channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Tue Jun 27 03:54:58 MST 2006


Author: oej
Date: Tue Jun 27 05:54:57 2006
New Revision: 36171

URL: http://svn.digium.com/view/asterisk?rev=36171&view=rev
Log:
Don't change direction of the dialogue when we send a re-invite
(will confuse to/from headers and to/from tags)

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=36171&r1=36170&r2=36171&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jun 27 05:54:57 2006
@@ -6100,13 +6100,10 @@
 	/* Use this as the basis */
 	initialize_initreq(p, &req);
 	p->lastinvite = p->ocseq;
-	ast_set_flag(&p->flags[0], SIP_OUTGOING);
 	return send_request(p, &req, 1, p->ocseq);
 }
 
-/*--- transmit_reinvite_with_t38_sdp: Transmit reinvite with T38 SDP ---*/
-/*     A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
-       INVITE that opened the SIP dialogue
+/*! \brief Transmit reinvite with T38 SDP 
        We reinvite so that the T38 processing can take place.
        SIP Signalling stays with * in the path.
 */
@@ -6125,7 +6122,6 @@
 	/* Use this as the basis */
 	initialize_initreq(p, &req);
 	p->lastinvite = p->ocseq;
-	ast_set_flag(&p->flags[0], SIP_OUTGOING);
 	return send_request(p, &req, 1, p->ocseq);
 }
 



More information about the asterisk-commits mailing list