The Panasonic SIP IP Phones are top-notch end points from a brand name you trust. Getting them setup and working usually isn’t very difficult if you know what info needs to go into what fields. The following tutorial applies to the follow phones, however, some interfaces may change slightly from model to model: KX-UT113-B, KX-UT123-B, KX-UT133-B, KX-UT136-B, KX-UT248, KX-UT670, KX-TGP500, KX-TGP550.
First you will need to setup the extension on your Asterisk System (the process can vary depending on the distribution). From this process, you will need the:
- IP Address of the Asterisk Server: ie: 192.168.42.100
- SIP Port of the Asterisk Server: ie: 5060
- Extension SIP ID/Alias: ie: MAC Address of phone
- Extension Number: ie: 7000
- Extension Password: ie: 1234pass
- Voicemail Access Number: ie: 8500
Next we will get started on configuring the phone itself. Please note, this is tutorial on how to get your phone connected. Advanced functionality is not covered:
- Connect the phone to the network and power it on
- Once connected to the network, you will need to turn on embedded web from the individual phone settings (turns off after a while). Press the Setup button > Select Embedded web > Select On
- Open up a web browser of your choice and type in the IP Address of the Panasonic IP Phone into the address bar. A login screen should appear. The default login is L: admin P: adminpass.
- Once logged in, you should make sure you have upgraded the firmware to the current version, the link to the admin guide and firmware files can be found here: http://www.panasonic.net/pcc/support/sipphone/download/us.html
- Select the VoIP tab and click on Line 1 under SIP Settings. Fill in these fields:
- Phone Number: Extension # assigned in Asterisk. ie 7000
- SIP URI: Extension SIP ID/Alias assigned in Asterisk. ie: MAC Address of phone
- Registrar Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
- Registrar Server Port: SIP Port of the Asterisk Server. ie: 5060
- Proxy Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
- Proxy Server Port: SIP Port of the Asterisk Server. ie: 5060
- Presence Server Adress: IP Address of the Asterisk Server. ie: 192.168.42.100
- Presence Server Port: SIP Port of the Asterisk Server. ie: 5060
- Outbound Proxy Server Address: IP Address of the Asterisk Server: ie: 192.168.42.100
- Outbound Proxy Server Port: SIP Port of the Asterisk Server: ie: 5060
- SIP Authentication ID: Extension SIP ID/Alias on the Asterisk. ie: MAC Address of phone
- SIP Authentication Password: Extension Password on the Asterisk. ie: 1234pass
Lastly, Select the Telephone tab. Then Select Line 1 under Call Control. Fill in these fields: Voice Mail Access Number: Voicemail Access Number on the Asterisk: ie: 8500
Click Save and reboot the phone.
Your Pansonic IP Phone should be configured to work on your Asterisk IP PBX. Any question? Just leave a comment or contact us at The Telecom Spot for assistance.
Music on hold do not work property.
If you answer an incomming call with panasonic phone, and hold it, the first time caller do not get music on hold sound. but if you recover call and hold, in second hold caller get music.
How is it?
Hi German, that’s hard to say without more specific information. What system or hosted platform are you using, also what phone and music-on-hold? Thanks.
I used these settings to try and connect a panasonic kx-tgp550T04 to a switchvox 65 cabinet and the phone remains in the voip status of “registering” and does not complete the process. Any ideas?
Assuming the extension in Switchvox is created, you may want to set your own “phone password” under “phone settings > advanced options” to test and see if that is what’s causing the problem. When manually entering the SIP credentials into the Panasonic, make sure the pass is exactly as you typed it (and the other information as well). Hopefully that will solve the problem for you.