Hi,
We are setting up a PBX with Asterisk and FreePBX.
We created 2 extensions (one for a Cisco IP phone and one for an X-Lite soft phone)
We creates 2 SIP trunks because we have 2 providers: one with one CID (CID1) and two channels and another one with 2 CIDs and 2 channels.
Everything works fine. We can put outgoing calls and answer incoming calls.
The problem we have is that all our clients call CID1 to joins us. We would like to redirect calls arriving to CID1 to the SIP accounts of our second provider in order to be able to receive calls until all our 4 channels are busy and then only redirect to a vocal message.
Any help greatly appreciated!!!
Thanks,
Gilles