Dummy SIP Trunks - Help

Hi Guys,

I have searched and searched and cannot find any information on how to do this… is it even possible?

I am looking to create a dummy SIP provider so that when I forward calls from a dialler I am using, it can transfer internally. At the moment I have to transfer it to an external number, which is ok, but can cause problems. I need to specify a SIP provider for the internal transfers to work (to group extensions etc…). I have tested this using a secondary Asterisk box and it works fine, however I don’t want to running.

Is there a way that I can create a ‘dummy’ SIP provider/trunk? It would call extensions/groups on the box itself, (localhost setup).

Any help would be much appreciated as my brain has given up on this now and the internet isn’t providing me many answers :frowning:

Many Thanks,

I am pretty sure your problem arises due to constraints imposed outside of Asterisk (e.g. an Asterisk GUI).

If not, you probably need to provide your dialplans and attempted dialplans, so that we can understand what you are trying to do and if or why a constraint exists.

you can use Local:

define a context in extensions.conf

exten => _0.,1,DoThis…

and later, in another context:


havent tested it recently, but that should work ! let us know

Hi Guys,

Thank you so much for your help.

My colleague also suggested to try the ‘Local’ as the SIP trunk and it did actually work.

My fear was that we wouldn’t be able to log an Agent in using Local as it wouldn’t know where it is going, appears to be fine though. Shall be testing tomorrow morning on live site.

Will keep you updated!