Hi all, New to the list. We are going to begin testing various voip gateways. I am trying to understand the reference to * in this thread. Is there a rule of the list that I need to be aware of ? Do not want to breech the etiquette of the list.
Thanks Frank ----- Original Message ----- From: "Gavin Hollinger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 02:58 Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > haven't been able to get it to pass dtmf to *. I don't know if this > > Do you have > dtmfmode=inband > in sip.conf? > > http://www.sippstar.com/en/631927444894185.html > > Q.: DTMF generated by SIPPS is not recognized by other > applications. > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > application awaiting an event instead of a tone will not be able to work > with SIPPS > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
