Hi, Here is the configuration of the server that I currently have
extension 100 (SIP) =>(SIP)asterisk server 1.8.18(IAX trunk) <===>(IAX trunk)asterisk server 1.4.32(SIP) ===> SIP Providers
The issue is while dialing out from extension 100(sip) if the providers sends back 180 Rining the SIP extension(100) won't hear the ringback tone, where as if the providers send 183 session in progress extension(100) will hear the ring back tone.
I tested registering to the main gateway server (by passing iax trunk) and it plays ring back tone every time for 180 Rining and for 183 session progress.
Any help would be highly appreciated. Thanks -- _____________________________________________________________________ -- 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
