Further to a recent post about a problem whereby the server continues to spew packets to the phone after hangup (sometimes, not every time), I have found that this problem appears to be alleviated by using RFC2833 instead of SIP INFO, however in switching to RFC2833 I introduce another problem - that DTMF tones for navigating menus become unreliable.
RFC2833 and SIP INFO are the only 2 options supported by the phone. Inbetween the phone and the server is the internet and 1 NAT traversal. Any ideas? Michael _______________________________________________ -- 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
