[OpenSIPS-Users] Register Free Opensips/Asterisk Integration

[Schneur Rosenberg]

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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