[OpenSIPS-Users] OpenSIPS+asterisk: cannot place call

[Stefano Sasso]

2010/9/27 Stefano Sasso <stesasso at gmail.com>:
>> if (is_method("INVITE")) {
>>        if (!ds_is_in_list("$si", "$sp")) ) /* if it's not from asterisk */
>>                if (!ds_select_dst("1", "5")) {
>>            ....
>>
>>        } else
>>                route(1); /* send it out */
>
> I inserted what you said, but now the destination rings, but the
> source has the call interrupted.

I also tried with the following configuration.

route{

	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}


	if (ds_is_in_list("$si", "$sp")) {
		xlog("RELAYING from dispatcher, method:$rm");
		t_relay();
		exit;
	}


	xlog("SIP route info: [$fu/$tu/$ru/$ci] METHOD: $rm");
	

	if (!has_totag()) {
		# initial request
		record_route();
	}
	else {
		# sequential request - obey the indicated route
		loose_route();
		t_relay();
		exit;
	}

	# CANCEL processing
	if (is_method("CANCEL"))
	{
		if (t_check_trans())
			t_relay();
		exit;
	}


	xlog("First route:
[USER:$au/CALL-ID:$ci/FROM-TAG:$ft/FROM-URI:$fu/FROM-URI-USER:$fU/METHOD:$rm]");

	if (is_method("REGISTER") || is_method("OPTIONS") ||
is_method("INVITE") || is_method("ACK")) {
		xlog("- dispatching METHOD $rm");
		if (!ds_select_dst("1", "5")) {
			xlog("Service unavailable for ds_select_dst[$rm]");
			send_reply("503","Service Unavailable");
			exit;
		}
	}
	
	else {
		send_reply("405","Method Not Allowed");
		exit;
	}

	# route the request
	if (!t_relay()) {
		sl_reply_error();
	}
}


This time I cannot place the call, and asterisk is saying:
SIP/2002-0000000a is circuit-busy :(

any other hint?
I worked on it the whole day without getting it working :(


-- 
Stefano Sasso
http://stefano.dscnet.org/