[Asterisk-Users] Stutter Tone

[Anton Krall]

True. I remember it was working on time but cant remember what config it
had.

Anybody using Granstreams handytone 286 atas? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven
Critchfield
Sent: Viernes, 04 de Marzo de 2005 09:26 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Stutter Tone

On Fri, 2005-03-04 at 21:10 -0600, Anton Krall wrote:
> I think I have something misconfigured regarding voicemails. They work 
> great, I have this setup:
> 
> Sip.conf
> 
> [ext1]
> Context=phones
> Mailbox=201
> 
> Voicemail.conf
> 
> [home]
> 
> 201,password,name,email at mail
> 
> Voicemail delivery and all works great but when I check sip extension 
> ext1 (analog phone using a Granstream ATA 286), the stutter tone 
> signaling message waiting does not work.

SIP dialtones come from the SIP device. Look up the config on your SIP
device.
--
Steven Critchfield <critch at basesys.com>

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