<p dir="ltr">Sure does, works perfectly.</p>
<div class="gmail_quote">On Mar 11, 2013 8:04 PM, "Nick Khamis" <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thank you so much for your responses!<br>
<br>
Schneur, I know that we were working on the similar architectures at<br>
some point, and had the same questions starting up. With your<br>
approach, do you still have the answering machine functionality<br>
defined by Asterisk (e.g., exten => _1XXX,1,Dial(SIP/${EXTEN}, 20))?<br>
<br>
Thanks in Advance,<br>
<br>
Nick.<br>
<br>
<br>
<br>
<br>
<br>
On 3/11/13, Schneur Rosenberg <<a href="mailto:rosenberg11219@gmail.com">rosenberg11219@gmail.com</a>> wrote:<br>
> I have a similar setup and I use the full URI for incoming calls, so<br>
> lets say the OpenSIPS server is at <a href="http://sip1.mycarrier.com" target="_blank">sip1.mycarrier.com</a> and I want to<br>
> send the call to a sip user called 101 then I send the call to<br>
> <a href="mailto:101@sip1.mycarrier.com">101@sip1.mycarrier.com</a><br>
><br>
> On Sun, Mar 10, 2013 at 4:04 AM, Nick Khamis <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
>> Hello Everyone,<br>
>><br>
>> I have gone through a few really good tutorials from the OpenSIPS<br>
>> site, Asterisk resources etc.. The unanswered question (and final<br>
>> piece of our puzzle) is if it's possible to have a register free<br>
>> environment in an OpenSIPS/Asterisk integration. Most approaches have<br>
>> OpenSIPS relay the UA's REGISTER request to Asterisk which has<br>
>> "host=dynamic" set for the Friend/Peer and everything works as<br>
>> expected.<br>
>><br>
>> Where I run into problems is in Inbound calls. When I try to call the<br>
>> extension from a DID I am receiving "Unable to create channel of type<br>
>> 'SIP' (cause 20 - Unknown)". And rightfully so!<br>
>> Reason being:<br>
>><br>
>> SIP Show Peers Yields:<br>
>><br>
>> Name/username Host Dyn Forcerport ACL Port<br>
>> Status Realtime<br>
>> 1001/1001 192.168.2.5 N 5060<br>
>> UNREACHABLE Cached RT<br>
>> TTrunk/<a href="http://sip.exp.com" target="_blank">sip.exp.com</a> 192.168.2.5 N 5060 UNKNOWN Cached<br>
>> RT<br>
>><br>
>><br>
>> As for who will keep track of the UA location, the OpenSIPS `location`<br>
>> table has the correct<br>
>> info:<br>
>><br>
>> select username,domain,contact,socket from location;<br>
>> +----------+--------------------+----------------------------+----------------------+<br>
>> | username | domain | contact | socket<br>
>> |<br>
>> +----------+--------------------+----------------------------+----------------------+<br>
>> | 1001 | <a href="http://sip.exp.com" target="_blank">sip.exp.com</a> | <a href="http://sip:1001@192.168.2.11:5060" target="_blank">sip:1001@192.168.2.11:5060</a> |<br>
>> udp:<a href="http://192.168.2.5:5060" target="_blank">192.168.2.5:5060</a> |<br>
>> +----------+--------------------+----------------------------+----------------------+<br>
>><br>
>> OpenSIPS: <a href="http://sip.exp.com" target="_blank">sip.exp.com</a><br>
>> OpenSIPS: 192.168.2.5<br>
>> Asterisk: 192.168.2.10<br>
>> UA: 192.168.2.11<br>
>><br>
>> I have set `host=<a href="http://sip.exp.com" target="_blank">sip.exp.com</a>' for the UA but the UA is still<br>
>> `UNREACHABLE` by asterisk<br>
>><br>
>> As for the rest of the media related stuff, everything works<br>
>> perfectly. Outbound works fine. As you know, this only poses a problem<br>
>> with inbound calls to the UAs.<br>
>><br>
>> Your Help is Greatly Appreciated,<br>
>><br>
>> Nick.<br>
>><br>
>> _______________________________________________<br>
>> Users mailing list<br>
>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
><br>
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><br>
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</blockquote></div>