Asterisk and SIP


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,


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

Reply to
Ramon F Herrera
Loading thread data ...


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?


Reply to
Glenn Robinson

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.


Reply to
Ivor Jones Forums website is not affiliated with any of the manufacturers or service providers discussed here. All logos and trade names are the property of their respective owners.