Below is a quick start guide for getting a Cisco/Linksys SIP handset up and running with Asterisk/FreePBX.
Both Linksys & Cisco phones have almost identical web admin setup pages but the layout and design differ slightly, setup procedures are identical for both.
Firstly plug the phone into the network via cat5 network cable (If you have 2 switch ports beneath the phone you want to use the port marked “SW”, don’t bother routing through the PC…it wont work well) and connect the power supply and plug in.
The phones get configured via a web interface, to do this you must first know the IP address of the phone. Shown below.
Assuming that you have got the IP address correct you will see the summary page of your phone.
Look at the top right hand corner of the screen
This will give you access to all the configuration options available.
Only change what this document says, you don’t need to change anything else.
This is where you set your network settings, ALLWAYS SET THE PHONE WITH A STATIC IP.
Change the following.
There is only one setting to change here, “RTP packet size”, the default setting for this is “0.030” it needs to be changed to “0.020”.
This page allows you to set the phone to your regional preference for dial tones, engaged tones, time and date etc., you only need to change 2 settings on this page
Reorder tone change from
Interdigit Long Timer Change from “10” to “5”
This is where you need to setup the extension that you wish the phone to use, to do this you must first have:
First click on “EXT 1” Tab, please note you can have more than one if need be.
Finally, really important for multiple phones on the same local network, they must transmit out through different ports, start with 5060 and work your way up. i.e. 5060, 5061 etc for each phone you setup.