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

[Kristian Kielhofner]

On Tue, Feb 3, 2009 at 3:49 PM, Joshua Colp <jcolp at digium.com> wrote:
>
> 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?
>

Josh,

  Thank you for the clarification.  I will spend some time today
looking at this.

  One issue (that I'm sure you're aware of):

RFC 2833:
3.6 Sending Event Packets

   An audio source SHOULD start transmitting event packets as soon as it
   recognizes an event and every 50 ms thereafter or the packet interval
   for the audio codec used for this session, if known.

  Regardless of bridging method used, Asterisk "SHOULD" start
transmitting even packets as soon as they are recognized.  After that
Asterisk does increment properly (even though Sonus does not) ;).

  Anyways, thank you, thank you, thank you for all of your help with this!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com