[asterisk-users] Loss of RTP stream during DTMF collection

[Dave George]

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