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SIP Trunks from the GUI?

Posted by mattdarnell on Mon, 05/19/2008

We are trying to allow another Asterisk box to use us for dialtone.

We are able to create the trunks, but when they place calls we do not send them to our SIP trunks. Our system keeps the call local. Does anyone know how to accept calls on a SIP trunk and connect them to another sip trunk automatically?

We also need to route DID's to the other asterisk box.

Thanks,
Matt


Submitted by eeman on Tue, 05/20/2008 Permalink

what context are you using? can you copy/paste the sip config for the specific trunk?

if you are doing asterisk-to-asterisk why not use IAX2 instead of sip?

Submitted by mattdarnell on Tue, 05/20/2008 Permalink

I chose SIP because I can make out the setup messages better than I can make out the IAX2 messages. No technical reason though. I think once we get the solution nailed most of the customers will be SIP based.

Here is the config...it is a connection from a Mitel 3300 but the issue is the same. When they try to dial out through my box, I don't relay it on to the PSTN.

[ComTel]

qualify=no

nat=no

;=description=ComTel's Mitel 3300

host=67.53.xxx.xxx

dtmfmode=rfc2833

context=from-outside

type=friend

canreinvite=no

disallow=all

allow=ulaw

allow=alaw

allow=gsm

Thanks,

Matt

Submitted by mattdarnell on Tue, 05/20/2008 Permalink

For this type of call:

PSTN --> Thirdlane --> IP-PBX --> Telephone

It would be nice to have an inbound route action like:

8085551212 ----> 5551212@66.34.223.102

That would send calls to another PBX and massage the DNIS digits.

-Matt

Submitted by eeman on Tue, 05/20/2008 Permalink

from-outside doesnt have dial patterns to dial back outside. You either need to pick a different context with dialing patterns OR you need to choose from-inside so they can use your existing dialing patterns.