<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 href="mailto:anca@opensips.org">anca@opensips.org</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; 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 href="http://www.voice-system.ro/" target="_blank">www.voice-system.ro</a></pre></font>
<div>
<div></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>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 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 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></div></div></div><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
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