I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio.
I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? Dave -- _____________________________________________________________________ -- 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
