[asterisk-users] Polycom Soundstation Conferencing Unit

[Danny Nicholas]

I never get that one right :(
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-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Novack
Sent: Wednesday, January 20, 2010 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Soundstation Conferencing Unit

FXS port is correct answer.
FXO ports are for PSTN or PBX lines
FXS SUPPLIES battery and ringing, and receives DTMF ( or pulse ) dialing
FXO receives battery and supplies DTMF ( or pulse ) dialing



Danny Nicholas wrote:
>
> According to what I see on Ebay, it is an Analogue handset. You would 
> have to hook it to an FXO port.
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> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Will Payne
> *Sent:* Wednesday, January 20, 2010 8:57 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Polycom Soundstation Conferencing Unit
>
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> On 20 Jan 2010, at 14:39, Dean Collins wrote:
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> Is it a SIP handset or analog style unit (or worse proprietary).
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> I'd say analogue.
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