[OpenSIPS-Users] Register Free Opensips/Asterisk Integration

[Nick Khamis]

Thank you so much for your responses!

Schneur, I know that we were working on the similar architectures at
some point, and had the same questions starting up. With your
approach, do you still have the answering machine functionality
defined by Asterisk (e.g., exten => _1XXX,1,Dial(SIP/${EXTEN}, 20))?

Thanks in Advance,

Nick.





On 3/11/13, Schneur Rosenberg <rosenberg11219 at gmail.com> wrote:
> I have a similar setup and I use the full URI for incoming calls, so
> lets say the OpenSIPS server is at sip1.mycarrier.com and I want to
> send the call to a sip user called 101 then I send the call to
> 101 at sip1.mycarrier.com
>
> On Sun, Mar 10, 2013 at 4:04 AM, Nick Khamis <symack at gmail.com> wrote:
>> Hello Everyone,
>>
>> I have gone through a few really good tutorials from the OpenSIPS
>> site, Asterisk resources etc.. The unanswered question (and final
>> piece of our puzzle) is if it's possible to have a register free
>> environment in an OpenSIPS/Asterisk integration. Most approaches have
>> OpenSIPS relay the UA's REGISTER request to Asterisk which has
>> "host=dynamic" set for the Friend/Peer and everything works as
>> expected.
>>
>> Where I run into problems is in Inbound calls. When I try to call the
>> extension from a DID I am receiving "Unable to create channel of type
>> 'SIP' (cause 20 - Unknown)". And rightfully so!
>> Reason being:
>>
>> SIP Show Peers Yields:
>>
>> Name/username     Host            Dyn    Forcerport ACL Port
>> Status               Realtime
>> 1001/1001              192.168.2.5            N      5060
>> UNREACHABLE Cached RT
>> TTrunk/sip.exp.com 192.168.2.5            N      5060     UNKNOWN Cached
>> RT
>>
>>
>> As for who will keep track of the UA location, the OpenSIPS `location`
>> table has the correct
>> info:
>>
>> select username,domain,contact,socket from location;
>> +----------+--------------------+----------------------------+----------------------+
>> | username | domain             | contact                    | socket
>>              |
>> +----------+--------------------+----------------------------+----------------------+
>> | 1001     | sip.exp.com | sip:1001 at 192.168.2.11:5060 |
>> udp:192.168.2.5:5060 |
>> +----------+--------------------+----------------------------+----------------------+
>>
>> OpenSIPS: sip.exp.com
>> OpenSIPS: 192.168.2.5
>> Asterisk: 192.168.2.10
>> UA: 192.168.2.11
>>
>> I have set `host=sip.exp.com' for the UA but the UA is still
>> `UNREACHABLE` by asterisk
>>
>> As for the rest of the media related stuff, everything works
>> perfectly. Outbound works fine. As you know, this only poses a problem
>> with inbound calls to the UAs.
>>
>> Your Help is Greatly Appreciated,
>>
>> Nick.
>>
>> _______________________________________________
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>> Users at lists.opensips.org
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>
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