on asterisk 1.4 I have 3 SIP trunks defined (001,002 and 003). When I get an incoming call on 001 I first try to use 002 to route the call outside (SIP provider) and then I try 003 (SIP/ISDN gateway).
Is there a way to mark one channel of the trunk 003 (which has a maximum of two channels) as used as soon as I detect the incoming call on 001 ?
Otherwise there's a possibility that if I have two outgoing calls on 001 (all channels busy) AND both calls are being processed (slowly) by the SIP provider I still can get external calls and those can block the internal calls from going out through 003.
Sorry for the messy explanation :)