[Asterisk-Users] SNOM 200 question

[YO Internet Information]

Are you sure that the snom isn't negotiating the GSM codec? I think that
this is negotiated by default unless you have disallow/allow statement. To
determine whether this is the problem,  put the following into the [general]
section of you sip.conf:

disallow=all
allow=ulaw
allow=alaw

As for the choppy sound on VM messages, i don't think you can do much about
this. It's more down to the design than anything. Try putting the call on
mute when listening to messages.

Hope that helps.
Tan
telappliant.com


----- Original Message ----- 
From: "Lane Hoskins" <lane at automatedhorizons.net>
To: <asterisk-users at lists.digium.com>
Sent: Friday, January 30, 2004 9:17 PM
Subject: [Asterisk-Users] SNOM 200 question


Question for other snom 200 users:

1. We have horrible sound quality regardless of the codec we use in the
phone or specify in *. Has anyone else run into this early on and found
a software fix?

2. Speakerphone will not work for playing VM messages, it chops the
message into unintelligible fragments of audio. Any ideas?

3. Initially we have horrible introduction of background noise into the
handset earpiece which seems to quiet after there is audio on the other
end. Ideas?

4. Sound quality to called parties outside our system is intermittently
horrible: static filled and raspy where we have to ask people to repeat
themselves many times. Could this be related to powerline noise or
something like that?


We have 8 lines coming into our building. Two are the main lines which
we have ringing to the receptionist first and then to selected other
extens. This part works great. We need to map the keys on the SNOM 200
such that when there is a call on line 1 the top key flashes/lights
steady depending on call state and any extension can pick it up even if
it doesn't ring there by pressing the button. This needs to hold true
for the 1st two lines, and one of the remaining 6 lines at each
extension as we have direct dials.

All calls come to * via a T1 Digium card and an Adtran TSU 600. There
are 8 separate POTS lines to our building for voice.

So in example - call comes in on pstn line 1 , button one flashes at all
phones, someone answeres it, button one solid on all phones, call comes
in on line 2, button 2 flashes on all phones,can be answered from
anywhere by simply hitting that button, gets answered and button changes
to solid on all phones, call comes to me from line 8 (my direct dial
line) and button 3 flashes on my phone only (my phone will also ring b/c
it's set up that way in the dialplan) I can put other caller on hold and
answer line 8 simply by pressing the button.

Is this an easy thing to do that I'm simply not seeing?


Thanks,

Lane Hoskins, MCP
Network Engineer
Automated Horizons Inc.
Direct - 540.767.7626
Main - 540.767.7600



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