I have a n00b problem... call "forwarded" with DID # and Asterisk?

Hello! I have installed Asterisk on a Ubuntu box, my “show core version” returns this from CLI:

Asterisk 13.11.2 built by asterisk @ jserve on a x86_64 running Linux on 2016-10-22 17:48:48 UTC

I have two “test” DID numbers and I would like to learn how to do this VERY SIMPLE process…

A user takes their cell phone and dials one of my test DID numbers, pretend, 17274204201. This then starts ringing my personal number, 17272660666, allowing me to answer them and have a conversation.

My problem is I have very limited test usage of these DID numbers and do not want to waste them with some errant configuration or something.

I’ve configured my asterisk IAX.conf and sip.conf to what my DID provider has recommended on their website. I’m fairly certain that information is accurate…

That leaves me with the extensions.conf file…


There is an example I used, and I barely tested it, enough locally to dial my own number from it and get a busy signal, I guess indicating that it worked.

What does the extensions.conf file look like, as an example, for having my DID number, 17274204201 “forward” the call to 17272660666?

Any help is appreciated. I’m sorry this is such a n00b problem. If you’d like to link the proper documentation or tutorials that might help me out, I’d much appreciate it. I’m attempting to use a fairly common DID number provider and I’d bugger them about this, but this problem is really just related to me not understanding asterisk.

Thanks in advance!

It would be Dial(SIP/17272660666@).

Do I need something behind that @? What is if I do not know where the phone is provided by? Sorry if this seems a silly question.

Right now I’m also debating on if I need a registrar for voip or not… and a good company for buying DID. I’ve looked at a few and haven’t really found a good one yet :(. any suggestions? Or, I’m not sure if that is allowed here.

I’ve actually become VERY familiar with all the conf files now, and just got my SIP stuff to work on my external ip, which was a lot harder than I thought it would be.

Thanks, btw :slight_smile:

Ugh, yes, discourse ate part of it. An example: Dial(SIP/17272660666@provider) where provider is the same of the configured ITSP in sip.conf

I don’t really have any suggestions as for the rest. Others would have more over all experience with various providers.

I think part of the issue here is that I would not know the provider in most instances. I think the work around is using something like Google Voice as my “trunk” to be able to handle the outbound dialing…

exten => _NXXNXXXXXX,1,Dial(Motif/jack/1${EXTEN}@voice.google.com,r)

Stuff like that works so far, once I connect something like linphone or X-Lite, I’m at least able to call my own GV number, so I’m hacking away at it now to see what else I can get it to do :).

My only problem I see in the future, is I don’t think I can use a single Google Voice number or account for handling many calls, especially at once…

Yes, you have to pay an upstream ITSP to connect to the telephony network and to provide calling to destinations.

Google voice is a provider.

Google Voice is working fine so far :smiley: haha; I’m just not sure how many calls it can handle at once…

You can not use Google Voice for anything except normal residential usage. They track usage patterns and if they determine anything other they WILL terminate your ability to use it.

Is there a way to pay them for some expanded type of service? I’m trying to Google that now…

I have no idea, but Google Voice itself is an unsupported service. It may disappear soon for example. It’s also not really supported by us for that reason. So you can use it - but your results may vary.

Ah, okay, what is a well supported and either free (unlikely) or very cheap trunk to use for the purposes I have outlined?

Anything that uses SIP.