- CANCEL never uses authentication

- Add docs on canreinvite

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 933b62cac0578916fb0199961cdc504bc3194735..2d380e9f17c6456e5a33618d6130cfe4821a9e94 100644 (file)
--- 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 169941e08266bd0675d641694328c137d894f235..eeb29a18e96f4f0063778c1aaea7821a7e55606d 100644 (file)
--- 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