On 22/04/05, Ian Hailey <asterisk@dinplug.com>
wrote:> Hello everyone,
>
> I am trying to receive DTMF commands on asterisk from PSTN calls
> terminated at my asterisk box. I have tried to terminate the PSTN calls
> with both SIP and IAX using sigate.co.uk and voipuser as the PSTN
> terminator. When I listen to tones sent from the PSTN side (e.g.
> continuous DTMF tone of about 3 seconds) on the asterisk server (stored
> in the voice mail) the tone is more or less completely muted, just the
> initial tone start can be heard. I am using the G711 codec. Does anyone
> have any idea if these tones are on purpose muted by the service
> providers or any other reason why it does not work?
I'm not aware of the detailed reason, but DTMF into Asterisk from
Sipgate won't work. This path is well-trodden...
http://www.voipuser.org/forum_topic_844.html amongst other places.
Peter
--
Peter Bowyer
Email: peter@bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter@bowyer.org