I have two Callcentric accounts, and want incoming calls from them to have different values for dtmfmode and directmedia. I tried setting up two different SIP Trunks in FreePBX, one for each account, but its not working. Calls from either account randomly goes to one SIP Trunk or the other depending on the IP address Callcentric is sending from.
My peer details for one of the accounts look like:
context=hpk-cid-zap-custom
fromuser=1777xxxxxxx
host=callcentric.com
fromdomain=callcentric.com
insecure=port,invite
secret=yyyyyyy
type=peer
defaultuser=1777xxxxxxx
dtmfmode=inband
qualify=yes
disallow=all
allow=ulaw
nat=yes
canreinvite=no
directmedia=no
srvlookup=yes
Is there some way to get the calls directed to the correct SIP Trunk or is there some other way to do this with the configuration files in /etc/asterisk?