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Hi Prem,<br>
<br>
Yes, your scenario is possible with OpenSIPS B2BUA and it will work
also with TCP/TLS. <br>
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
<a class="moz-txt-link-freetext" href="http://www.opensips.org/Resources/B2buaTutorial#toc15">http://www.opensips.org/Resources/B2buaTutorial#toc15</a>).<br>
<br>
Regards,<br>
<pre class="moz-signature" cols="72">--
Anca Vamanu
<a class="moz-txt-link-abbreviated" href="http://www.voice-system.ro">www.voice-system.ro</a></pre>
<br>
<br>
On 07/26/2010 02:38 PM, Premalatha Kuppan wrote:
<blockquote
cite="mid:AANLkTinm06X+rFwTPEANjTK=+5PqWkN=2VcM1tJo1CGb@mail.gmail.com"
type="cite">
<div>Thanks Anca.</div>
<div> </div>
<div>Iam going to try this now and let you the results.</div>
<div> </div>
<div>BTW, i have one question:</div>
<div> </div>
<div>Iam using Opensips 1..6.2 TLS enabled, can i make it to works as
well as B2BUA ? If so how ?</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>In my case, Destination address is either TLS/TCP/UDP enabled.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>Please advice.</div>
<div> </div>
<div> </div>
<div>Thanks,</div>
<div>Prem<br>
<br>
</div>
<div class="gmail_quote">On Mon, Jul 26, 2010 at 3:50 PM, Anca Vamanu
<span dir="ltr"><<a moz-do-not-send="true"
href="mailto:anca@opensips.org">anca@opensips.org</a>></span> wrote:<br>
<blockquote
style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"
class="gmail_quote">
<div bgcolor="#ffffff" text="#000000">Hi Prem,<br>
<br>
There are some things that you have to take care if you want TCP to
work:<br>
<br>
1) set the script parameter : tcp_accept_aliases = 1'<br>
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.<br>
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.<br>
<br>
Regards,<br>
<font color="#888888">
<pre cols="72">--
Anca Vamanu
<a moz-do-not-send="true" href="http://www.voice-system.ro/"
target="_blank">www.voice-system.ro</a></pre>
</font>
<div>
<div class="h5"><br>
<br>
On 07/26/2010 08:54 AM, Premalatha Kuppan wrote:
<blockquote type="cite">Hi,<br>
<br>
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.<br>
<br>
Below is the Error observed at Opensips.<br>
<br>
Jul 23 13:54:45 204548-4 /usr/local/sbin/opensips[8374]
<div>: new branch at <a moz-do-not-send="true">sip:999_456_1000@209.242.149.98:1036;transport=TCP</a><br>
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s<br>
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed<br>
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcp_send: connect failed<br>
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:tm:msg_send: tcp_send failed<br>
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:tm:t_forward_nonack: sending request failed<br>
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=<a
moz-do-not-send="true"
href="mailto:6ef81f740f13e4d778513d1315fc76fc@204.12.57.221"
target="_blank">6ef81f740f13e4d778513d1315fc76fc@204.12.57.221</a>;code=477;reason=Request
Failure<br>
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=<a
moz-do-not-send="true"
href="mailto:11504845-3488896563-937560@sg2-lax.yourvoip.net"
target="_blank">11504845-3488896563-937560@sg2-lax.yourvoip.net</a>;code=200;reason=OK<br>
<br>
Any Insight ?<br>
<br>
Thanks,<br>
Prem</div>
</blockquote>
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