Hi Bogdan,<br><br>This the logic i used to handle calls from asterisk to opensips. But i thnk since there would be via header, call is going to destination via Asterisk (which doesn't support TCP/TLS)<br><br>Can you please help. I want the call to be handled by opensips after IVR done by asterisk.<br>
<br>i.e finally callee<---------------------->opensips<------------------>called<br><br>What needs to adapted ?<br><br> if ($rU==NULL) {<br> # request with no Username in RURI<br> sl_send_reply("484","Address Incomplete");<br>
exit;<br> }<br> <br> if (is_method("INVITE"))<br> if (method=="INVITE|BYE")<br> {<br> $var(z)=$(tu{uri.user});<br> if($var(z)=~"[0-9]+_[0-9]+_[0-9]+") {<br>
xlog("Call from Asterisk PROTOCOL=$pr PORT=$op PROTOCOL PORT=$oP RQT=$oU \n");<br> if (!lookup("location")) {<br> sl_send_reply("404","Not here"); }<br>
}<br> }<br><br>Please advice.<br><br>Thanks,<br>Prem<br><br><div class="gmail_quote">On Fri, Jul 23, 2010 at 11:31 AM, Premalatha Kuppan <span dir="ltr"><<a href="mailto:premalatha@ngintech.com">premalatha@ngintech.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Bogdan,<br><br>Thanks for your reply. Is the logic correct ?<br><br>After IVR, i want asterisk to transfer call to opensips and opensips to handle the call further. But, i still wonder how opensips would get the call details of orignator, do i need to manipulate nything.<br>
<br>Also, after googling found Opensips1.6x is compactable with 1.4x. So Iam using the same versions. But, yesterday i tried insalling asterisk 1.6x (since it has TLS/TCP) support). So far successful, i can forward calls to asterisk from Opensips. Will I get any other problems..i dono..Please advice.<br>
<br>I appreciate your valuable comments.<br><br>Thanks,<br><font color="#888888">Premalatha</font><div><div></div><div class="h5"><br><br><br><br><br><div class="gmail_quote">On Thu, Jul 22, 2010 at 8:18 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Premalatha,<br>
<br>
have you checked the opensips logs for error? can you post a call flow<br>
showing exactly which step fails ?<br>
<br>
Regards,<br>
<font color="#888888">Bogdan<br>
</font><div><div></div><div><br>
<br>
Premalatha Kuppan wrote:<br>
> Hi,<br>
><br>
> I posted before my query; but no repsonse :(<br>
><br>
> Can some1 helps, whether this logic works,<br>
><br>
> Iam using opensips 1.6.2(TLS) and Asterisk(1.4.3.1).<br>
><br>
> 1. all users registering @ opensips<br>
> 2. All calls to opensips forwarded to sasterisk for IVR; thorugh IVR<br>
> input destination is known.<br>
> 3. Since asterisk 1.4x version doesnt support TLS/TCP. Iam forwarding<br>
> call to Opensips; indeed here some messy happens. Asterisk sents<br>
> invite to Opensips, opensips reach the destination via asterisk (First<br>
> of all, iam not sure this logic is true). For TLS, call is successful<br>
> but no audio. For TCP, call fails after IVR.<br>
><br>
> Only my destination is TLS/TCP enabled.<br>
><br>
> I appreicate valuable input and advice on thsi logic.<br>
><br>
> Please help.<br>
><br>
> Thanks,<br>
> Prem<br>
</div></div><div><div></div><div>> ------------------------------------------------------------------------<br>
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<br>
--<br>
Bogdan-Andrei Iancu<br>
OpenSIPS Bootcamp<br>
20 - 24 September 2010, Frankfurt, Germany<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
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