On 22/04/05, Ian Hailey <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
