Asterisk and SIP

Sep 16, 2005 2 Replies

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:

formatting link
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.

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


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

Join the Discussion

Have something to add? Share your thoughts — no account required.

Didn't find your answer?

Ask the community — no account required