[Asterisk-Users] Stutter Tone

dean collins dean at collins.net.pr
Fri Mar 4 22:33:36 MST 2005


The grandstream 101's have stutter tone.

Works great with asterisk at home.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Howard
Lowndes
Sent: Friday, March 04, 2005 11:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Stutter Tone

On Sat, 2005-03-05 at 14:10, 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.
> 
> Anything wrong with contexts or something?

Stutter tome works with Zap but does it work with SIP phones unless they
have their own stutter tome which activates when they get a "NEW
MESSAGE" header.  My HOP 1002 SIP phones certainly don't have that.


> 
> Thx Guys
> 
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-- 
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
------------------------------------------
"Flatter government, not fatter government;
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