WLAN600 Wireless IP Phone

When I received the WLAN600 phone and took it out of its package, I was not impressed with its construction. To me the phone felt if I where to drop it, it would shatter.  However, the phone did come with a belt clip, a docking charger, AC adapter and user guide. 

Setting up the WiFi and SIP was not difficult at all.  For the WiFi just follow the instruction in the book.  For SIP configuration I captured the screens that I made changes to.  See below:

WLAN600-login.jpg (14655 bytes) To configure the phone you will need to first obtain the IP address of the phone, then point your browser to the IP address.  Once done, you should get the login password pop up.  See image at left.  Enter " voipadmin" in username, " admin" for password (both without the quotes") and click OK.

SIP.CONF

[72466]
type=friend
username=72466
secret=1234
host=dynamic
context=voice-mail
dtmfmode=rfc2833
allow=g729
canreivet=no
mailbox=72466@local

EXTENSION.CONF

[home]
exten => _724xx,1,Answer
exten => _724xx,2,Dial(SIP/${EXTEN},15,t)
exten => _724xx,3,Voicemail(u${EXTEN})
xten =>   _724xx,4,Hangup
exten => _724xx,103,Voicemail(b${EXTEN})
exten => _724xx,104,Hangup

VOICEMAIL.CONF

[local]
74266 => 74266,Kurt Pasewaldt,kurt@pasewaldt.com

 
WLAN600-main.jpg (77294 bytes) The actions above will bring you to the main screen.  Click on "SIP Proxy".
WLAN600-sip.jpg (96458 bytes) In the first field "SIP URI" I put the extension of the phone and the IP address of the Asterisk server.  I left the port to the default setting of 5060.
The Second field "SIP Server Address" is the IP address of the Asterisk Server.
Leave the third field at 5060.
"REGISTER Server Address" is the IP address of the Asterisk server and the port is 5060.  The next three fields I left with the default options. 
"Display Name" I changed to the the extension  and for the Authentication I used the extension of the phone.  So, "REGISTER Username" is the extension and the password is whatever you want.
WLAN600-dsp.jpg (85270 bytes) After configuring the SIP Proxy I clicked on "DSP Setting".  The phone only has two CODECs to select from: G711a/ulaw and G729 8k.  I recommend that you use G729 CODEC because when the phone is using G711 it sometimes  thinks it is running G729. This bug can cause one-way audio, no audio, or choppy voice when configured for G711a/ulaw.  NOTE: You need a license to run G729 on Asterisk.
I left the "Speaking" and "Listening" fields at the defaults and changed the "RTP Port" to 10,000.  The two radio buttons I left at "Large", selected "inband(RFC2833)" and a payload of 101. 
I then applied the configuration and restarted the phone.