On Tue, 30 Dec 2003, Tilghman Lesher wrote: > On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote: > > On Thu, 18 Dec 2003, Aaron Martin wrote: > > > I have upgraded my grandstream phone from firmware 1.0.3.78 to > > > 10.0.4.30 and now I am having problems with early dial. On the > > > older firmware earlydial worked fine with my asterisk server, but > > > now as soon as I have dialed the number I get a congested tone, and > > > the number 4 flashes up on the LCD screen. > > > > > > Has anyone had this problem, and if so, how do I fix it? > > > > Early dial has never worked for me, and I just upgraded to the > > 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, > > making it impossible to check my voice mail. > > > > As an example, my extension is "100" and let's say my password is > > "1234". Here is what * captures: > > > > -- Executing VoiceMailMain("SIP/damin-3099", "") in new stack > > -- Playing 'vm-login' (language 'en') > > NOTICE[5126]: File chan_sip.c, Line 4667 (handle_response): Peer > > 'damin' is now REACHABLE! > > -- Playing 'vm-password' (language 'en') > > -- Incorrect password '1112234444' for user '11000' (context = > > <any>) -- Playing 'vm-incorrect' (language 'en') > > > > Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will > > yield better results. > > What happens when you change the configuration of the GS phone to > send DTMF via SIP INFO?
I had that set originally. I get the same behavior no matter wether I use "Send via SIP, RTP or INLINE AUDIO". -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users