Cisco 7940

Summary: Fantastic phone, the best speakerphone ever, however at least some features are broken in the older software releases (6.x)

Introduction

The phone is familiar to anyone who has walked through a modern office. However, what you see in an office is usually driven by Cisco Call Manager and not SIP. While it is possible to use the Asterisk skinny protocol implementation and run the phone "natively" I have never tried it. I first started using it before I had asterisk so I had to update it to SIP anyway.

Flashing the Phone

The security of this phone is laughable. Passwords or other protection can be broken in a matter of seconds because the phone will check for new firmware and new config via tftp at every boot. All you need is a linux box to serve the correct settings. So if you have a phone from a dot.bomb crater you should follow the following procedure:

• Obtain SIP OS binary for the phone. How, where and when - your problem. I can rant for a while on CSCO support of open standards as exemplified by their pricing of SIP firmware for the 79XX series.
• Set up tftp on your linux server and make sure it works
• Check if the phone has a static IP address in the menu
  • If the phone has a static IP address set up its gateway address and call manager address and as aliases on your linux box. This way the phone will query the box when booting for new firmware
  • If the phone has a dynamic address it will query the tftp server for firmware and config so there is nothing to do there
• Put the firmware in the /var/lib/tftpboot directory
• Put a file called OS79XX.TXT with the name of your firmware file(s) without the extension(s) in /var/lib/tftpboot on your server.
• Put a file called SIPDefault.cnf in /var/lib/tftpboot. Mos t of the settings can be obtained off CCO, however the following is the "important ones"

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)

Voila - your phone no longer has a password and is running a SIP load.

Asterisk Configuration

The phone works reliably vs any asterisk I have tried with the following caveats

• VAD does not work properly because the phone does not generate any keep-alive/noise comfort frames at least in release 6. As a result the line gets dropped after 20-30 seconds if the phone is put on mute

-- AntonIvanov? - 26 Nov 2008