[OpenSIPS-Users] TCP CONNECT ERROR;

[Premalatha Kuppan]

Thanks Anca. This is what i was looking for.

Btw, if i understood correct. I can make current Opensips (TLS enabled -
1.6.2 (Proxy)) one to act as B2BUA. or is it i hv to uninstall the current
one and install other 2BUA. Iam confused. Please clarify.



On Mon, Jul 26, 2010 at 6:15 PM, Anca Vamanu <anca at opensips.org> wrote:

> 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 Vamanuwww.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> 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 Vamanuwww.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;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;code=200;reason=OK
>>
>> Any Insight ?
>>
>> Thanks,
>> Prem
>>
>>
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