Re: [Users] Disconnect Cause on OpenSER - sr-users

6 Jul 2006

On 7/5/06, raviprakash sunkara &lt;sunkara.raviprakash.feb14(a)gmail.com&gt; wrote:
...

 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 &lt;bogdan(a)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(a)openser.org [mailto:users-bounces@openser.org] On  Behalf
 Of Mike Williams
Sent: Wednesday, July 05, 2006 8:04 PM
To: users(a)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

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

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