Asterisk and SIP session refresh with UPDATE request
Posted on Wed 15 February 2012 in VoIP
Using directmedia=update or (canreinvite=update in old-sytle configuration) in chan_sip.conf Asterisk sends SIP UPDATE request to refresh SIP session.
If this UPDATE requests refers to an inactive call the phone reply with SIP/2.0 481 Call Leg/Transaction Does Not Exist.
In this situation Asterisk continue in resending UPDATE and doesn't tear
down the previous allocated channel.
Using INVITE instead UPDATE (directmedia=yes) Asterisk correctly
terminate the dialog.
This little chan_sip.c patch (Asterisk 1.8.9.1 version) correct this behavior:
--- asterisk-1.8.9.1/channels/chan_sip.c 2012-02-06 22:40:37.000000000 +0100
+++ /root/asterisk-1.8.9.1/channels/chan_sip.c 2012-02-08 15:30:04.552673312 +0100
@@ -20952,6 +20952,17 @@ static void handle_response(struct sip_p
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
+ }
+ else if (sipmethod == SIP_UPDATE) {
+ switch (resp) {
+ case 481: /* Call leg does not exist */
+ ast_log(LOG_WARNING, "Re-invite with UPDATE to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
+ //xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
+ if (p->owner)
+ ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ break;
+ }
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
You can download the patch at https://github.com/pbertera/junk/blob/master/asterisk-stuff/asterisk-1.8.9.1-UPDATE-reinvite.patch
Asterisk reported issue: ASTERISK-19313