[asterisk-users] Sipgate > DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

[Kristian Kielhofner]

On Wed, Jan 13, 2010 at 2:43 AM, listuser at spamomania.co.uk
<listuser at spamomania.co.uk> wrote:
> Thanks for that. Looking at the RTP packets I can see two types as you
> point out. The first appears to be the audio:
>
> Real-Time Transport Protocol
> 10.. .... = Version: RFC 1889 Version (2)
> Payload type: ITU-T G.711 PCMU (0)
>
> And as you say, the DTMF events are clear to see:
> RFC 2833 RTP Event
> Event ID: DTMF One 1 (1)
> ..00 1010 = Volume: 10
>
> So, as these can be seen in the stream, do I need to tell Asterisk to
> detect these? Does it not do that when I set: dtmfmode=rfc2833
> ???

  There are some pretty widely recognized RFC2833 compatibility issues
in the SIP/RTP world.  Which version of Asterisk are you using?  Do
you know what kind of equipment your carrier is using?  If they are
using Asterisk you can try to add rfc2833compensate=yes to their peer
entry in sip.conf.

>>
>> The SIP debug, however, will tell you if the remote end is configured
>> to use RFC2833 or not.  That's why I was telling you to look for
>> telephone-event in the INVITE from your provider.  Keep in mind SIP
>> (most likely) runs over UDP between you and your provider, not TCP.
>>
> I see a 'telephone-event' :
>
> a=rtpmap:101 telephone-event/8000
>

  That's all you need to know.  They are configured for RFC2833 and
they're sending RFC2833.

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