[asterisk-commits] oej: branch 1.4 r47733 - in /branches/1.4: channels/ configs/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Thu Nov 16 09:03:49 2006
New Revision: 47733

URL: http://svn.digium.com/view/asterisk?view=rev&rev=47733
Log:
- CANCEL is never authenticated (according to the RFC)
- Update docs on canreinvite. "nonat" is the recommended setting for most users with
  phones behind a NAT.

Modified:
    branches/1.4/channels/chan_sip.c
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=47733&r1=47732&r2=47733
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Nov 16 09:03:49 2006
@@ -3304,7 +3304,7 @@
 					/* Do we need a timer here if we don't hear from them at all? */
 				} else {
 					/* Send a new request: CANCEL */
-					transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
+					transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
 					/* Actually don't destroy us yet, wait for the 487 on our original 
 					   INVITE, but do set an autodestruct just in case we never get it. */
 					needdestroy = 0;
@@ -11444,11 +11444,11 @@
 	if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
 		/* if we can't BYE, then this is really a pending CANCEL */
 		if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
-			transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
+			transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
 			/* Actually don't destroy us yet, wait for the 487 on our original 
 			   INVITE, but do set an autodestruct just in case we never get it. */
 		else 
-			transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+			transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
 		ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);	
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 	} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=47733&r1=47732&r2=47733
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Thu Nov 16 09:03:49 2006
@@ -248,6 +248,12 @@
 				; route = Assume NAT, don't send rport 
 				; (work around more UNIDEN bugs)
 
+;----------------------------------- MEDIA HANDLING --------------------------------
+; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; no reason for Asterisk to stay in the media path, the media will be redirected.
+; This does not really work with in the case where Asterisk is outside and have
+; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
+;
 ;canreinvite=yes		; Asterisk by default tries to redirect the
 				; RTP media stream (audio) to go directly from
 				; the caller to the callee.  Some devices do not