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 [s@default:1] Playback("SIP/1647XXXXXXX-b75331a8", "vm-goodbye") in new stack
-- <SIP/1647XXXXXXX-b75331a8> Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [s@default: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 also have FreePhoneLine and wanted to setup the asterix server on my NAS. My installation was done without any problems (installed version 1.4 with GUI 2.0). However, when I try to connect to the FPL, the status messages say’s unregistered. So for some reason I can’t register to FPL server. Now I see that you were succesful in connecting yours. Could you by any chance indicate step by step procedure as to how you set your TRUNK up? which .conf file did you edit and what exacly did you put in there to connect?
ie. in which file did you put in the following details:
Moreover, I don’t understand what you mean by registeration string 1647XXXXXXX:PASSWORD@voip.freephoneline.ca? is this necessary to include and if so where do you put it?