I have a ZyXel Prestige P-2000W V2 handset which I am using with an Asterisk PBX. I am putting this page on the web because I ran into quite a bit of difficulty making the phone work with my PBX. I am using firmware version WV.00.01 with Asterisk version 1.0.7 (the Debian 3.1 package).

I am able to do the following:

• Make calls by phone number through a SIP proxy (the Asterisk PBX)
• Receive calls via the Asterisk PBX
• Receive numerical Caller ID on incoming calls (names are displayed if the number is in the phone's directory)
• Use arbitrary WiFi networks with arbitrary NAT settings to place and receive calls

I am not able to roam from network to network without dropping a call in progress. I do not use peer to peer calling, so I do not know whether it works. All of my calls are via the Asterisk PBX.

Here is my SIP configuration, annotated to indicate which values in sip.conf correspond to which fields in the phone's web configuration interface.

[--your username--]                     ; Username portion of the "SIP URI" field and "Registrar Username"
type=friend
context=--your outbound context--
username=phone
host=dynamic
nat=yes
qualify=no
dtmfmode=rfc2833			; Set "DTMF Relay" to "outband"
disallow=all
allow=ulaw				; Set "Default Voice Codec" to "G.711u, 64k"
secret=--your password--		; "Registrar Password"

On the phone, you will also need to set these additional values:

• Set "Network Mode" to "DHCP"
• Set "SIP Server Address" and "Registrar Server Address" to either the DNS name or IP address of your Asterisk PBX
• On the "NAT TRVSL" form, set "Select Type" to "STUN (RFC3489)"
• Set "STUN Server IP" to the DNS name or IP address of your STUN server. I am using the public server at stun01.sipphone.com

Happy calling...