I'm running asterisk 1.6.2.6 and freepbx-2.7.0
My trunk is configured as follows:
Outgoing Settings
Trunk name: GoTalk
Peer Details:
host=sip.gotalk.com
username=09xxxxxx
secret=YNxxxxxx
type=peer
fromuser=09xxxxxx
fromdomain=sip.gotalk.com
canreinvite=no
insecure=very
Incoming Settings
User Context: 09xxxxx
User Details:
username=09xxxxx
fromuser=09xxxxx
type=peer
secret=YNxxxxx
insecure=very
host=dynamic
fromdomain=sip.gotalk.com
context=from-pstn
Register String:
09xxxxxx:[email protected]/09xxxxxx
I have an inbound route called Incoming with DID 09xxxxxx diverted to local extension 200
When I do a sip trace and dial my telephone number 0741xxxxx I just get failure beeps. I never see any SIP traffic from GoTalk to my asterisk server trying to connect the call.
Seems I'm not registering correctly for incoming calls because GoTalk aren't sending them to me. I am correct in using the GoTalk username 09xxxxxx as the DID, aren't I ? I've tried using my phone number but it makes no difference.