Cannot direct inbound SIP calls

So I am trying to receive inbound calls from freephoneline.ca. I have setup an inbound route (catch all route: any DID/ any CID).

I have also setup a trunk with the registration string: 1647XXXXXXX:[email protected], the following ougoing details:

Trunk Name: FPLTrunk
PEER Details:
username=1647XXXXXXX
type=friend
secret=PASSWORD
qualify=no
insecure=very
host=voip.freephoneline.ca
fromdomain=voip.freephoneline.ca
disallow=all
canreinvite=yes
allow=ulaw&alaw&gsm&g729
nat=yes

and the following incomming details:

USER Context: 1647XXXXXXX
USER Details:
disallow=all
allow=ulaw&alaw&gsm&g729
canredirect=no
context=from-trunk
fromdomain=voip.freephoneline.ca
fromuser=1647XXXXXXX
secret=PASSWORD
type=friend
username=1647XXXXXXX
nat=yes

The problem is that when dialing in, instead of connecting to the inbound route, the call gets transfered to the [default] context, the server plays the goodby tune and then the call gets disconnected. Here is the traceback from asterisk:

– Executing [[email protected]:1] Playback(“SIP/1647XXXXXXX-b75331a8”, “vm-goodbye”) in new stack
– <SIP/1647XXXXXXX-b75331a8> Playing ‘vm-goodbye.gsm’ (language ‘en’)
– Executing [[email protected]:2] Macro(“SIP/1647XXXXXXX-b75331a8”, “hangupcall”) in new stack

Now, I am at a loss and any help would be deeply appreciated …

Forgot to say that when simulating incomming calls from the PBX, then the incomming route is activating with no problem, which is really odd …

I don’t see a ‘context=from-pstn’ or equivalent in your trunk setup…

I lost perhaps 6 hrs because of this --> once I added it, it worked like a charm !!!

Tried to use the DID field, but apparently that does not work very well …

Tried also fiddling with the registration string:

1647XXXXXXX:[email protected]/1647XXXXXXX

That did not work either. I am all out of ideas :slight_smile:

Well, it solved by itself :slight_smile: Now it works …

And… what did you do to solved it ?