int gotdest;
const char *p_replaces;
char *replace_id = NULL;
+ int refer_locked = 0;
const char *required;
unsigned int required_profile = 0;
struct ast_channel *c = NULL; /* New channel */
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
transmit_response(p, "482 Loop Detected", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/*! This is a spiral. What we need to do is to just change the outgoing INVITE
* so that it now routes to the new Request URI. Since we created the INVITE ourselves
*/
ast_string_field_set(p->owner, call_forward, peerorhost);
ast_queue_control(p->owner, AST_CONTROL_BUSY);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
}
transmit_response_reliable(p, "491 Request Pending", req);
ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
/* Don't destroy dialog here */
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
}
ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
/* Do not destroy existing call */
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
if (sipdebug)
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
/* Todo: (When we find phones that support this)
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
+ } else {
+ refer_locked = 1;
}
/* The matched call is the call from the transferer to Asterisk .
ast_channel_unlock(p->refer->refer_call->owner);
}
}
+ refer_locked = 0;
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
transmit_response_reliable(p, "488 Not acceptable here", req);
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
} else {
res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
if (res == AUTH_CHALLENGE_SENT) {
p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
if (res < 0) { /* Something failed in authentication */
if (res == AUTH_FAKE_AUTH) {
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_set(p, theirtag, NULL);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
/* If T38 is needed but not present, then make it magically appear */
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_debug(1, "No compatible codecs for this SIP call.\n");
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
if (p->rtp) {
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
}
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
gotdest = get_destination(p, NULL); /* Get destination right away */
change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
p->invitestate = INV_COMPLETED;
update_call_counter(p, DEC_CALL_LIMIT);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/* If no extension was specified, use the s one */
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
p->stimer->st_active_peer_ua = TRUE;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
break;
ast_hangup(c);
sip_pvt_lock(p); /* pvt is expected to remain locked on return, so re-lock it */
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/* Go and take over the target call */
if (sipdebug)
ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
- return handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+ res = handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+ refer_locked = 0;
+ goto request_invite_cleanup;
}
}
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
+
+request_invite_cleanup:
+
+ if (refer_locked && p->refer && p->refer->refer_call) {
+ sip_pvt_unlock(p->refer->refer_call);
+ if (p->refer->refer_call->owner) {
+ ast_channel_unlock(p->refer->refer_call->owner);
+ }
+ }
+
return res;
}