Hi Paul, Happy Friday back.
In the config of the extension change the dtmf=XXX Basically there are three ways dtmf can be transmitted by a sip handset, choose another or search the voip-info for the options and you'll solve your problem pretty quickly. Re: sipgate....sorry cant help, you'll need to provide more info. Cheers, Dean > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Paul A Brown > Sent: Friday, 11 August 2006 9:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Problem with dtmf and voice mail > > Hi Guys, > > Happy Friday > > I have 2 problems.... > > I run [EMAIL PROTECTED] with some Cisco 7960's > > 1) DTMF - When I dial a number on the 7960 it works fine. However if I dial > a number that asks 'Dial 1 for this and 2 for that' and I hit 1 or 2 (or > whatever0 the other end acts as though nothing is heard. Any ideas? > > 2) Voicemail - I use a company called sipgate for my internal route. When > someone calls from outsied the call never goes to vmail. However if I dial > from ext to ext it does... > > Any ideas? > > Thanks > > Paul > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
