[Serusers] Strange problem - SIP-->PSTN - 40 sec calls duration

[deviator]

Thanks for reply!

This is part of my openser.cfg

..........

route {

	# ------------------------------------------------------------------------
	# Record Route Section
	# ------------------------------------------------------------------------
	if (method=="INVITE" && client_nat_test("3")) {
		setflag(7);
		record_route_preset("212.212.212.212:5060;nat=yes");
	} else if (method!="REGISTER") {
		record_route_preset("212.212.212.212:5060");
	};

......

	# ------------------------------------------------------------------------
	# Message Handler Logic
	# ------------------------------------------------------------------------
	if (loose_route()) {
		append_hf("P-hint: Loose Routed\r\n");
		if (has_totag() && (method=="INVITE" || method=="ACK")) {
			if (isflagset(7) || search("^Route:.*;nat=yes")) {
				setflag(6);
				use_media_proxy();
			};
		};
		route(1);
		break;
	};
	if (uri!=myself) {
		append_hf("P-hint: External Destination\r\n");
		route(1);
		break;
	};
	if (uri==myself) {
		append_hf("P-hint: Local Destination\r\n");
		if (method=="ACK") {
			setflag(1);
			route(9);
			break;
		} else if (method=="CANCEL") {
			route(5);
			break;
		} else if (method=="INVITE") {
			setflag(1);
			route(5);
			break;
		} else if (method=="REFER") {
			route(5);
			break;
		} else if (method=="REGISTER") {
			setflag(1);
			route(3);
			break;
		} else if (method=="OPTIONS") {
			options_reply();
			break;
		} else if (method=="SUBSCRIBE") {
			route(4);
			break;
		};
		lookup("aliases");
		if (uri!=myself) {
			append_hf("P-hint: Alias External Destination\r\n");
			route(1);
			break;
		};
		if (!lookup("location")) {
			sl_send_reply("404", "User Not Found");
			break;
		};
	};
	append_hf("P-hint: USRLOC Applied\r\n");
        route(1);
}

route[1] {
	remove_hf("Proxy-Authorization");
	t_on_reply("1");
	if (!t_relay()) {
		if (method=="INVITE" || method=="ACK") {
			end_media_session();
		};
		sl_reply_error();
	};
}
route[2] {
	# ------------------------------------------------------------------------
	# Call Forwarding Reply Route Handler
	# ------------------------------------------------------------------------
	if (!lookup("location")) {
		rewritehost("195.135.204.85"); # PSTN GW IP ADDRESSS GOES HERE
	} else {
		route(8);
		route(1);
	};
}

...........
...........
...........

route[7] {

	# ------------------------------------------------------------------------
	# PSTN Handler
	# ------------------------------------------------------------------------

	rewritehost("195.135.204.85");	# PSTN GW IP ADDRESSS GOES HERE
	if (method!="CANCEL") {
		if (!proxy_authorize("", "subscriber")) {
			proxy_challenge("", "1");
			break;
		};
		consume_credentials();
		avp_write("i:45", "inv_timeout");
		route(8);
	};
	t_on_failure("1");
	route(1);
}

route[8] {
	if (isflagset(6) || isflagset(7)) {
		use_media_proxy();
	};
}

........

onreply_route[1] {

	if (isflagset(6) || isflagset(7) || search("212.212.212.212")) {
		if (status=~"(180)|(183)|2[0-9][0-9]") {
			if (!search("^Content-Length:\ +0")) {
				append_hf("P-hint: NATed Reply\r\n");
				use_media_proxy();	
			};
		};
	};
	if (client_nat_test("1")) {
		fix_nated_contact();
	};
} 

.........

I dont think that my UA is broken, same results have all my ipphones and softphones :(

-----Original Message-----
From: Andrei Pelinescu-Onciul <andrei at iptel.org>
To: deviator <deviator at inbox.ru>
Date: Wed, 2 Nov 2005 19:58:06 +0100
Subject: Re: [Serusers] Strange problem - SIP-->PSTN - 40 sec calls duration

> 
> On Nov 02, 2005 at 18:48, deviator <deviator at inbox.ru> wrote:
> > Hello!
> > 
> > When i'm trying to call from UA to PSTN i have a call failure after 40 sec. All my UA's (private and public ip) works fine when calling each other. Calls from PSTN ---> UA's also works fine. But when i'm calling to PSTN i may talk only 40 sec. I'm tried from NAT UA's and UA's with public ip, softphones and ipphones - same result. I don't know what to do, someone please help me.
> 
> 
> The ACK doesn't reach the gateway. Do you drop it in your ser.cfg?
> Anyway the UA seems broken, it uses strict routing, but it does not copy
>  the entire uri from Record-Route:
> 
>  ACK sip:989099037259 at 212.212.212.212:5060;nat=yes
> 
> for Record-Route:
> <sip:989099037259 at 212.212.212.212:5060;nat=yes;ftag=2245526517777;lr=on>.
> 
> 
> (notice the missing lr=on in the ACK uri).
> 
> 
> Andrei
> 
> [...]
>