I configured the trunk and I can make outgoing calls.
However I can’t register my DID.
The only username I have is my DID.
I tried the following register string:
DID:password@server
And also this:
DID@server:password:DID@server:port/DID
But i won’t work.
The SIP provider says they do not provide help for ASTERISK / FreePBX.
They say that their user have reported figuring it out on their own but they don’t know how.
And of course the encourage to buy their crappy equipment instead.
Let me know if you have any idea of what I should try.
+1 to Dickos suggestion. That should help you figure out any errors. If outgoing works ok, but you cant receive calls i’m wondering if it may be more of a problem with incoming routes… I reckon along side debugging, you should set your trunk up as basically as possible, and create a very simple incoming route that captures any CID / DID and forwards to one extension… once that is working, start building complexity from there.
these basic settings should work on most SIP providers in the trunk:
in Outoing:
host=<Sip provider>
username=<Your Sip provider user>
secret=<Your sip provider password>
type=friend
nat=yes << change that to no if the server is directly connected to the net.
insecure=invite
qualify=yes
leave incoming blank
register string:
<sip user>:<sip password>@<sip server>/<sip user> or <sip user>:<sip password>@<sip server>
Then delete all your incoming routes, and create a new one ANY DID/CID and send to one of your extensions. Then as dicko suggested use sip set debug ip from asterisk CLI… If that doesn’t give you any useful info… you could also try core set verbose 10 which will announce everything that happens in asterisk