On Sat,Mar 03 09:17:PM, K bharathan wrote: >thru DSL connection the callers can hear only > one way;asterisk pbx is behind NAT;
Greetings, If the calls are esteblished,then SIP did its work. The system may RTP problems, NAT may or may not be the casue for the issue. Defaults RTP ports for asterisk are UDP 10000-20000, You may want to look into that direction. Guy Gold -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
