[asterisk-users] Register Free Opensips/Asterisk Integration

[Nick Khamis]

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.