SoundStation IP 3000

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Polycom Soundstation IP 3000

The Polycom Soundstation IP 3000 conferencing phone comes in at least two flavors, one is a standardized H.323 and the other is for 3com NBX systems. They are rumored to have different ROM chips so that the difference between them is more than a firmware update.

I tested the H.323 version with Asterisk and I was able to get it to work with the OH323 H.323 implementation. Initially I could only call from an Asterisk SIP phone to the IP 3000,. After updating the firmware in the IP 3000 I was able to call into Asterisk but it no longer worked the other way as it had before! This was using asterisk without a Gatekeeper.

Polycom now has a SIP conferencing phone available, the SoundStation IP 4000.



Avoid these if you can, but the OOH323 driver will work, except for in-call DTMF.

Note that the default password is not the usual Polycom '456'. For the IP 3000, the default administrator password is '**#'. With Asterisk IP as the first gateway IP, you can dial out with '0*xxx' where '0*' chooses the first gateway on the IP 3000 and 'xxx' is the extension in your Asterisk dialplan. Be sure port 1720 (H323) and your RTP range is open on your firewall.

Under trixbox, this phone extension would go under a custom extension like 'OOH323/192.168.1.54'

Here's a minimal /etc/asterisk/ooh323.conf

[general]
bindaddr=192.168.1.42
disallow=all
allow=ulaw
context=custom-crazyoldphone

[crazyoldphone]
type=friend
ip=192.168.1.54





Polycom Soundstation IP 3000

The Polycom Soundstation IP 3000 conferencing phone comes in at least two flavors, one is a standardized H.323 and the other is for 3com NBX systems. They are rumored to have different ROM chips so that the difference between them is more than a firmware update.

I tested the H.323 version with Asterisk and I was able to get it to work with the OH323 H.323 implementation. Initially I could only call from an Asterisk SIP phone to the IP 3000,. After updating the firmware in the IP 3000 I was able to call into Asterisk but it no longer worked the other way as it had before! This was using asterisk without a Gatekeeper.

Polycom now has a SIP conferencing phone available, the SoundStation IP 4000.



Avoid these if you can, but the OOH323 driver will work, except for in-call DTMF.

Note that the default password is not the usual Polycom '456'. For the IP 3000, the default administrator password is '**#'. With Asterisk IP as the first gateway IP, you can dial out with '0*xxx' where '0*' chooses the first gateway on the IP 3000 and 'xxx' is the extension in your Asterisk dialplan. Be sure port 1720 (H323) and your RTP range is open on your firewall.

Under trixbox, this phone extension would go under a custom extension like 'OOH323/192.168.1.54'

Here's a minimal /etc/asterisk/ooh323.conf

[general]
bindaddr=192.168.1.42
disallow=all
allow=ulaw
context=custom-crazyoldphone

[crazyoldphone]
type=friend
ip=192.168.1.54





Created by: smithnc, Last modification: Thu 21 of Jun, 2012 (07:44 UTC) by admin
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