10 mar 2013 kl. 03:04 skrev Nick Khamis sym...@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 HostDynForcerport ACL Port
Status Realtime
1001/1001 192.168.2.5N 5060
UNREACHABLE Cached RT
TTrunk/sip.exp.com 192.168.2.5N 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@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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