Hello, I am attempting to use Asterisk as a protocol converter.
I have set up asterisk to route incoming h323 calls to a SIP termination carrier. I make a test, call is coming correctly, is rerouted to termination carrier. Call connects and phone rings. Then, I pick up the phone and it hangs up after 2 seconds. I initially thought it was a codec issue. I made sure codec is g729 in all sip.conf & h323.conf parts (general context + specific contexts). Still, call drops after connects and gives error "cannot bridge between X call and Y call". Is this familiar to anyone? Do you have idea what to search next? _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
