[gballiet]

Hi all.

I have a sip account with Peopleline, a voip provider here in B.C. - they're based in Vancouver. I am having problems getting it to work with freepbx.
My sip.conf is as follows,

username=*********
type=peer
secret=xxxxxxxx
port=5062
host=sip.op.ucantalk.net
fromuser=**********
dtmfmode=rfc2833
disallow=all
context=from-sip-external
canredirect=no
allow=ulaw
allow=alaw

When I run sip show channels I get the following

Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
66.48.40.108 6046303772 796104ff358 00175/00000 unkn No

My output from sip debug is

<-- SIP read from 66.48.40.108:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 70.77.34.207:5060;branch=z9hG4bK3bcc7e53;rport=506 0;received=70.77.32.207
From: <sip:6046303772@gk.ucantalk.net>;tag=as0ed1ad21
To: <sip:6046303772@gk.ucantalk.net>;tag=2azxo54972
Call-ID: xxxxxxxxxxxxxxxxxxxxx@127.0.0.1
CSeq: 177 REGISTER
Contact: <sip:s@70.77.34.207>;expires=120
Date: Fri, 08 Dec 2006 18:24:52 GMT
Server: snom-proxy-linux/2.51
Content-Length: 0


So it looks to my untrained eye that I have registered and set up my trunk properly - but I can't get routing to work. My route selection works - but I get a busy signal from asterisk always. Any ideas?

Thanks in advance.

Gary