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- CANCEL is never authenticated (according to the RFC)

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- Update docs on canreinvite. "nonat" is the recommended setting for most users with
  phones behind a NAT.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-11-16 15:03:49 +00:00
parent e83e3123f7
commit 5bd53e3588
2 changed files with 9 additions and 3 deletions
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channels/chan_sip.c
+3 -3
View File
@@ -3304,7 +3304,7 @@ 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;
@@ -11444,11 +11444,11 @@ 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)) {
configs/sip.conf.sample
+6
View File
@@ -248,6 +248,12 @@ 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