From a427a2a89a03f8d32272504d14cea9da10a8095d Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Thu, 16 Nov 2006 15:12:30 +0000
Subject: [PATCH] - CANCEL never uses authentication - Add docs on canreinvite
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 6 +++---
configs/sip.conf.sample | 6 ++++++
2 files changed, 9 insertions(+), 3 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 933b62cac0..2d380e9f17 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3422,7 +3422,7 @@ static int sip_hangup(struct ast_channel *ast)
/* 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;
@@ -11647,11 +11647,11 @@ static void check_pendings(struct sip_pvt *p)
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)) {
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 169941e082..eeb29a18e9 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -266,6 +266,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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
--
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