Round 4 here for me, and hopefully this time I picked the right starting point. Now running my dedicated box on Ubuntu server, with Asterisk installed using apt-get, and asterisk-gui according the recipies floating around. All seems to be working at that level.
[quote=“the Asterisk-gui status screen”]OS Version:
Linux PhoneBox 2.6.28-11-server #42-Ubuntu SMP Fri Apr 17 02:45:36 UTC 2009 x86_64 GNU/Linux
Asterisk GUI-version : SVN-branch-2.0-r4980[/quote]
I’m on a Virgin Media (ex-NTL) cable modem, with an OpenWRT router between that and everything else. Ports 5004, 5060, 8000-8012 and 10000-20000 are forwarded direct to the Asterisk box.
My sipgate line is configured as a trunk, reported in Asterisk-gui as “Registered”, and by sipgate as “online”.
My Grandstream Budgetone 100 phone is set up as extension 6000, as a SIP/IAX user, with the IP of my server as its provider, and is reported by Asterisk-GUI as online and free.
I have an outgoing call rule of “0!” which (hopefully) allows any number with a zero at the start to be passed across to the trunk (with the removal of the first digit) - my intention here is “dial zero to get an outside line”.
I have an incoming calling rule with a pattern of “s” which (again, I hope) passes all inbound calls to extension 6000.
But I seem to have missed something. Inbound external calls to the sipgate number get an engaged tone, and dialing 010000 gives a 404. What I’d hoped for with 010000 was the standard sipgate test message “your phone is now configured correctly”.
Note that I haven’t delved into the conf files yet, all setup so far has been via the GUI. Hoping to keep it that way, am I being over-optimistic? I think that setting up the awful Nortel UNIstim-only phone might need me to get the spanners & blowtorch out, so I’ll leave that til last!
What did I miss?