On 02/19/10 18:38, Edwin Lam wrote: >> >> So I can only use one "context" for incoming calls. If I split the sip.conf >> into two files will it make any difference. > >there might be an "include" directive in sip.conf (i can't confirm) >however Asterisk will see it as one big sip.conf so it will do >absolutely nothing for you in this situation. > >what you can do is setup automatic dial to different extensions on >the 2 ports on audiocodes.
I already have setup automatic dialing, it does noting. But the solution might be to specify different port number in the Tel to IP routing table, and setup sip.conf entries to listen on these ports. Calls coming from Trunk Group 1 are to be sent on port 5065, and all calls coming from Trunk Group 2 will be sent on Trunk Group 5066. It will take two routing table entries to do this. -- Joseph -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
