[OpenSIPS-Users] Register Free Opensips/Asterisk Integration

[Schneur Rosenberg]

Sure does, works perfectly.
On Mar 11, 2013 8:04 PM, "Nick Khamis" <symack at gmail.com> wrote:

> 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.
> >>
> >> _______________________________________________
> >> Users mailing list
> >> Users at lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
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