[Asterisk-Users] SIP and DTMF

[Jeremy McNamara]

Scott England wrote:

> Being relatively new to * I has what may be a simple question, I haven't
> been able to find it in the archives though, or at least been able to
> recognize it.
>
> I have a 400P the is acting as a pstn gateway. It then forwards via IAX2
> to another * server at another site. The calls then get routed via
> callerid to a sip client with an exten  statement in extensions.conf.
> However I cant seem to get DTMF to forward to the sip extension. With
> IAX debugging I can see the dtmf call at both iax points but I dont see
> it happen at the sip point. Do I have to do something different then a
> simple exten => 6000,1,SIP/6000 at 192.168.0.108   ?
>

Make sure you are using RFC2833 dtmf mode.   INFO is another option if 
RFC2833 isn't supported by your sip device.



Jeremy McNamara