[asterisk-users] keep incoming codec same as outcoming on sip proxy
Hi, i have two computer with asterisk. One is a SIP proxy that Dial() the other. It is possible to be sure that the proxy does not make transcoding in any case and Hangup() the call if the Second asterisk does not support the codec ? Thanks ___ -- 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
[asterisk-users] keep incoming codec same as outgoing on sip proxy
Hi, i have two computer with asterisk. One is a SIP proxy that Dial() the other. It is possible to be sure that the proxy does not make transcoding in any case and Hangup() the call if the Second asterisk does not support the codec ? Thanks ___ -- 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
