[Asterisk-Users] Grandstream Early Dial

[Greg Boehnlein]

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".

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