<div class="gmail_quote"><div>Hi Bogdan,</div><div><br></div><div>Thanks for reply. I forgot to mention earlier that for I'm using OpenSIPS + FreeRadius, where radius is doing accounting and authentication. I used aaa_does_uri_exist() function as well, but seems not working or making mistake while implementing it. On other hand using lookup("location",m) function, on retcode = -1, I redirected the INVITE to GW, using Dispatcher. But though thanks for your suggestion and I'll consider it. </div>
<div><br></div><div>Few things I want to ask you, as I listed below;</div><div>1-How can I forward SIP INVITE request to other SIP machine in state full manner ?</div><div>2- While accounting using radius, when user A (registered on OpenSIPS) calls the user B who is located at GW side, accounting doesn't take place. On the other hand when user B (from GW) calls user A (to OpenSIPS), accounting take place. I want to know its cause? Because I want its accounting on both sides.</div>
<div><br></div><div>Kindly advise me at your earliest.</div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
------------------------------<br>
<br>
Message: 6<br>
Date: Thu, 18 Mar 2010 10:23:27 +0200<br>
From: Bogdan-Andrei Iancu <<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>><br>
Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS<br>
To: OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br>
Message-ID: <<a href="mailto:4BA1E2FF.3060702@voice-system.ro">4BA1E2FF.3060702@voice-system.ro</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
Hi Ahmed,<br>
<br>
if the destination number (called number) is not a local subscriber (a<br>
SIP user), you simply route the call to a PSTN GW (you do this re-route<br>
from the script)<br>
<br>
To check if a user is a local subscriber, you can either check a pattern<br>
(like all my local users are alphanumeric, or all starts with 3345*,<br>
etc), either simply check if the user does exists in the subscriber<br>
table (see the URI module, the db_does_uri_exists() function:<br>
<a href="http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131" target="_blank">http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131</a><br>
<br>
Regards,<br>
Bogdan<br>
<br>
Ahmed Munir wrote:<br>
> Hi,<br>
><br>
> I want to know how can I check the peers of source and destination<br>
> phones? Like if both phones are located (registered) on one<br>
> UAS(OpenSIPS) can call SIP-SIP, if any one phone is registered on UAS<br>
> and other is on PSTN, call will be re-routed to SIP-PSTN. In case of<br>
> SIP-SIP, lookup("location") function works and I need to know how can<br>
> I forward call to SIP-PSTN ?<br>
><br>
> Kindly advise me the method/ function can used for it.<br>
><br>
> --<br>
> Regards,<br>
><br>
> Ahmed Munir<br>
><br>
><br>
> ------------------------------------------------------------------------<br>
><br>
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><br>
<br>
<br>
--<br>
Bogdan-Andrei Iancu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
<br><br></blockquote></div>-- <br>Regards,<br><br>Ahmed Munir<br><br><br>