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

[Kevin P. Fleming]

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.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org