[OpenSIPS-Users] TCP CONNECT ERROR;

[Anca Vamanu]

Hi Prem,

Yes, your scenario is possible with OpenSIPS B2BUA and it will work also 
with TCP/TLS.
When asterisk gets the destination it can send a BYE to OpenSIPS with a 
header ( with any name you like) containing the destination. You just 
have to take care to put this name in the scenario ( modify this 
http://www.opensips.org/Resources/B2buaTutorial#toc15).

Regards,

-- 
Anca Vamanu
www.voice-system.ro



On 07/26/2010 02:38 PM, Premalatha Kuppan wrote:
> Thanks Anca.
> Iam going to try this now and let you the results.
> BTW, i have one question:
> Iam using Opensips 1..6.2 TLS enabled, can i make it to works as well 
> as B2BUA ? If so how ?
> Actaully, my scenario is All users will get registered with opensips 
> and all calls (Incoming Invite) would be forwarded to Asterisk 
> (1.4.3.1, it has no TLS, TCP support) for IVR. After IVR, asterisk 
> will get the destination address.
> In my case, Destination address is either TLS/TCP/UDP enabled.
> I was searching little bit, and found Opensips can handle Refere-to 
> header. From asterisk can i do transfer to Opensips and Opensips 
> should terminate the call log with asterisk and connect source and 
> destination party.
> Will this logic work and opensips has that capabilities? If so what 
> needs to be configured to make it as B2BUA with the existing setup.
> Please advice.
> Thanks,
> Prem
>
> On Mon, Jul 26, 2010 at 3:50 PM, Anca Vamanu <anca at opensips.org 
> <mailto:anca at opensips.org>> wrote:
>
>     Hi Prem,
>
>     There are some things that you have to take care if you want TCP
>     to work:
>
>     1) set the script parameter : tcp_accept_aliases = 1'
>     2) the top most Via header in Register from the client must
>     contain 'alias' parameter.  To check this print the Register
>     message from the script: xlog("$mb\n"); If you don't see this
>     parameter, you can fix it from the server by calling
>     force_tcp_alias() on that Register.
>     3) the ip and port  in the top most Via for Register must match
>     exactly the ip and port in the Contact header for tcp reusage to
>     work. Check this.
>
>     Regards,
>
>     -- 
>     Anca Vamanu
>     www.voice-system.ro  <http://www.voice-system.ro/>
>
>
>
>     On 07/26/2010 08:54 AM, Premalatha Kuppan wrote:
>>     Hi,
>>
>>     I have Integrated setup of Opensips(TLS) (1.6.2)) and Asterisk
>>     (1.4.3.1). When i try to make call to TCP enabled client. The
>>     call fails.
>>
>>     Below is the Error observed at Opensips.
>>
>>     Jul 23 13:54:45 204548-4 /usr/local/sbin/opensips[8374]
>>     : new branch at sip:999_456_1000 at 209.242.149.98:1036;transport=TCP
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
>>     ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
>>     ERROR:core:tcpconn_connect: tcp_blocking_connect failed
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
>>     ERROR:core:tcp_send: connect failed
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
>>     ERROR:tm:msg_send: tcp_send failed
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
>>     ERROR:tm:t_forward_nonack: sending request failed
>>     Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]: ACC:
>>     call missed:
>>     timestamp=1279907695;method=INVITE;from_tag=as2bf86f8d;to_tag=;call_id=6ef81f740f13e4d778513d1315fc76fc at 204.12.57.221
>>     <mailto:6ef81f740f13e4d778513d1315fc76fc at 204.12.57.221>;code=477;reason=Request
>>     Failure
>>     Jul 23 13:54:57 204548-4 /usr/local/sbin/opensips[8374]: ACC:
>>     transaction answered:
>>     timestamp=1279907697;method=BYE;from_tag=as7d156ab4;to_tag=3488896563-937568;call_id=11504845-3488896563-937560 at sg2-lax.yourvoip.net
>>     <mailto:11504845-3488896563-937560 at sg2-lax.yourvoip.net>;code=200;reason=OK
>>
>>     Any Insight ?
>>
>>     Thanks,
>>     Prem
>
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