[Asterisk-Users] Grandstream Early Dial
damin at nacs.net
Wed Dec 31 06:50:53 MST 2003
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 184.108.40.206 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
> > 220.127.116.11 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: 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".
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