My ata uses dtmf=info and my sip.conf uses dtmfmode=rfc2833. Do they have to match? Weird thing is, when making calls, transfer prompt works, but no for incoming.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Domingo, 20 de Marzo de 2005 03:56 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem transfering incoming calls looks like an dtmf mode setting problem, make sure you have it set to dtmfmode=rfc2833 or dtfmmode=info in sip.conf, the same goes for your ata. On Sun, 20 Mar 2005 15:29:18 -0600, Anton Krall <[EMAIL PROTECTED]> wrote: > Guys. > > Im having a big problem transfering incoming calls thru zap channels > to some other extension. If the call is made by me to the outside via > zap channels, no problem, hitting # gets me the transfer prompt, but > if the call comes in thru zap and eventhough I am sending the call > from the zap channel to my sip ata (GS ata 286) using Dial with wtWT > as parameters, when hitting # I don't hear the prompt. > > Any ideas what might be wrong? > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
