Asterisk and SIP

Hello,

I've set up asterisk and I can use it internally without any problems.

I've now got my self a phone number from Gradwell (uk) and have set up a
trunk in AMP and set incoming calls to my number to be routed through to my
softphone.

When I dial my new number from a PSTN phone I can see it logged in asterisk
with Disposition of "NO ANSWER". This is true as I get the standard "Your
call cannot be connected" message on the PSTN phone.

Any ideas why the calls are getting in to my asterisk server but then not
being routed through to my extension?

Thanks

 

Glenn:

In my experience, you are not going to get many answers to such
specific question in this NG.

Your best bets are: the Asterisk Group in Google, or this forum:

http://forums.digium.com/

Then, there is the mailing list(s).

Good luck,

-Ramon

ps: the first thing you are going to be asked is to post your sip.conf
and extensions.conf files.

 

You might want to try uk.telecom.voip as well, quite a few people there
know a lot about Asterisk (I'm not one of them..!). Unfortunately the
group is suffering from an attack of the trolls at the moment, but if you
can ignore them you might find some answers.

Ivor