On 05/25/2012 04:30 PM, Dave George wrote:
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?

You are missing quite a lot of crucial information required for anyone to help you. First, what version of Asterisk are you using? Second, what type of channel is being used to connect to Asterisk? You mention it works from a SIP phone, but not from a media gateway.. is that gateway also using SIP, or something else? What does 'during DTMF collection' mean? Do you mean after a prompt has been played and the voicemail application is waiting for input, or is this during prompt playback, or something else?

Quite some time ago Asterisk was changed to ensure that silence would be sent while an application was running and waiting for input from the caller; if your version is older than this, then that could explain what you are seeing. That's just a mildly-educated guess though.

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