asterisk users - Mar 2013 - Register Free Opensips/Asterisk Integration

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.

10 mar 2013 kl. 03:04 skrev Nick Khamis <symack at gmail.com>:
> 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.
> 
There are a lot of models for this. Check my presentation from Astricon
2010 to get some ideas.
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations

/O
> 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.
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users