This works fine in extensions.conf:
exten => _0X./100,1,Dial(SIP/[EMAIL PROTECTED])
exten => _0X./200,1,Dial(SIP/[EMAIL PROTECTED])
This will just use different SIP channels for different Caller ID's.
If I write the same to a realtime table, Asterisk always uses sipout-a, no matter what Caller ID is used.
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
