hello Bogdan,

Same problem occurred for me , but I'm Using X-lite 3.0 .
And i'm put  on DMZ in my router ( NETGEAR and BELkenn) .


I install the openser 1.0.1  and rtp proxy 0.3 in same  Linux System
openser server is located with public id xx.xxx.xxx.xx of   192.168.2.2 ,
And  UAC are outside the NAT, 
When  one UAC call to other UAC( are both in  outside the NAT where openser server), after the INVITE method get request by server,  after 32 second its hung up automatically, Voice is ok , and callee is hung upping, not caller,
UAC ( inside the nAT , openser server ) in not hung uping and voice is not ok....

 I think problem is not in NETWORK and it may in  RTp , NAT ,
Can u help on this ,
Where is the problem, in NAt with rtp or networking,
Here by Mime's openser.cfg
****************************************
route{
    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    };
    if (msg:len >=  2048 ) {
        sl_send_reply("513", "Message too big");
        exit;
    };

    # NAT detection
    route(2);

    if (!method=="REGISTER")
        record_route();

    if (loose_route()) {
        append_hf("P-hint: rr-enforced\r\n");
        route(1);
    };

    if (!uri==myself) {
        append_hf("P-hint: outbound\r\n");
        route(1);
    };

    if (uri==myself) {
        if (method=="REGISTER") {
            if (!www_authorize("xx.xxx.xxx.xxx", "subscriber")) {
                www_challenge("xx.xxx.xxx.xxx", "0");
                exit;
            };

            if (isflagset(5)) {
                setflag(6);
                # if you want OPTIONS natpings uncomment next
                # setflag(7);
            };
            save("location");
            exit;
        };

        if (!lookup("location")) {
            sl_send_reply("404", "Not Found");
            exit;
        };
        append_hf("P-hint: usrloc applied\r\n");
    };

    route(1);
}


route[1] {
    if (subst_uri('/(sip:.*);nat=yes/\1/')){
        setflag(6);
    };

    if (isflagset(5)||isflagset(6)) {
        route(3);
    }

    if (!t_relay()) {
        sl_reply_error();
    };
    exit;
}

route[2]{
    force_rport();
    if (nat_uac_test("19")) {
        if (method=="REGISTER") {
            fix_nated_register();
        } else {
            fix_nated_contact();
        };
        setflag(5);
    };
}

route[3] {
    if (is_method("BYE|CANCEL")) {
        unforce_rtp_proxy();
    } else if (is_method("INVITE")){
        force_rtp_proxy();
        t_on_failure("1");
    };
    if (isflagset(5))
        search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
    t_on_reply("1");
}

failure_route[1] {
    if (isflagset(6) || isflagset(5)) {
        unforce_rtp_proxy();
    }
}

onreply_route[1] {
    if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") {
        force_rtp_proxy();
    }
    search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');

    if (isflagset(6)) {
        fix_nated_contact();
    }
    exit;
}

On 7/5/06, Bogdan-Andrei Iancu < bogdan@voice-system.ro > wrote:

Hi,

it might be a signalling problem. Most of the UA drops the calls if they
do not get the ACK for 200 OK.
check on the network if this is the case.

regards,
bogdan

Hamid Ali Asgari wrote:

>The calls are between two UAs.
>The problem is that with a certain type of UA (type A), the calls are ok if
>the calls are between two type A UAs and don't get disconnected. I can talk
>as long as I want.
>
>But if I try calling from that UA (type A) to Windows messenger, the call
>gets disconnects after less than a minute. In the 1 minute I can talk (so I
>assume it's not a CODEC problem, correct me if I am wrong)
>
>I have also tried with a UA and a Cisco gateway. On the Cisco debugs I see
>Disconnet cause code 102 (Session-End-Callback ) which I don't think would
>be the case. There is no callback config on the gateway or the UA.
>
>I guess the UA is tearing down the call for some reasn I don't know.
>
>Any clues?
>Hamid
>
>
>-----Original Message-----
>From: users-bounces@openser.org [mailto: users-bounces@openser.org] On Behalf
>Of Mike Williams
>Sent: Wednesday, July 05, 2006 8:04 PM
>To: users@openser.org
>Subject: Re: [Users] Disconnect Cause on OpenSER
>
>On Wednesday 05 July 2006 12:31, Hamid Ali Asgari wrote:
>
>Are the calls from one UA to another, or from a UA to a gateway? I know for
>instance that Asterisk has problems with G729b silence detection and will
>drop calls because it thinks the call has dropped. Perhaps other equipment
>or
>carriers has this problem too.
>
>---Mike
>
>
>
>
>>Hi,
>>
>>I am having a problem with OpenSER and certain types of CPEs. The problem
>>is that the calls get established and the parties can talk, however after
>>
>>
>a
>
>
>>very short period the call gets disconnected. Any guidelines how I could
>>troubleshoot this?
>>
>>
>>
>>PS: Is there anyway to see the calls disconnect cause on OpenSER? I am
>>currently running OpenSER with radius.
>>
>>
>>
>>Thanks in advance,
>>
>>Hamid
>>
>>
>
>_______________________________________________
>Users mailing list
>Users@openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>_______________________________________________
>Users mailing list
>Users@openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>


_______________________________________________
Users mailing list
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http://openser.org/cgi-bin/mailman/listinfo/users

--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.
www.hyperion-tech.com
+91-9985077535