[asterisk-dev] Update: Asterisk DTMF issues with Sonus

[Joshua Colp]

----- "Kristian Kielhofner" <kristian.kielhofner at gmail.com> wrote:

> Hello everyone,
> 
>   It seems we have finally found the cause of all of our DTMF issues
> with Sonus equipment.
> 
>   When Asterisk is bridging two SIP channels with RFC 2833 DTMF it
> appears to wait for the end of the event from the transmitting device
> before re-transmitting the event to the other (bridged) channel.
> While Asterisk is waiting for the event to end it isn't transmitting
> RTP (or anything else) to the other device.  This can be seen as a
> gap
> in RTP traffic to the bridged channel in between the last voice frame
> and the start of the DTMF event.
> 

This depends on a few factors. If your two SIP channels are Packet2Packet bridged then the RTP packets coming in get immediately written out. If your two SIP channels are going through the core then the DTMF should start as soon as we get the begin from the other side and end when we get the end. If your two SIP channels are going through the core and DTMF features are enabled then it may be delayed. What is your configuration like and Dial options?

-- 
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org