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
I chose SIP because I can
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
For this type of call: PSTN
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
from-outside doesnt have
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.
Thanks Eric...I will try
Thanks Eric...I will try that.
Can I send you a pineapple or chocolate covered macadamia nuts?
-Matt
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?