Horas and konnichiwa untuk semua

I got time (and enough motivation) to finally got rid of the SCCP firmware inside this Cisco IP 7960 that has been idle for years.

1. Setup a TFTP server
2. Put these files on the TFTP server
OS79XX.TXT
*.bin
(E.g., for v6.3, the file is P0S3-06-3-00.bin)
3. Set 7960 for firmware update
Press **# at the main window of 7960 to unlock the phone, then set DHCP to yes, and alternate to TFTP to yes. Then, press "*","6",Setting buttons at the same time to reboot. The 7960 will then boot with SIP firmware
4. Put these files on the TFTP server
OS79XX.TXT
SIPDefault.cnf
dialplan.xml
RINGLIST.DAT
5. Set the 7960 for SIP setting
Press "*","6",Setting buttons at the same time to reboot again. The 7960 will then booth with SIP settings according to the contents of SIPDefault.cnf and SIP<MAC addresss>.cnf (for example, SIP000B46CABF9F.cnf if the MAC address of the 7960 is 000B46CABF9F)

References:
Configuring Cisco 79xx phones with Asterisk
Cisco 7960 IP Phone – SIP firmware version
Using Cisco IP Phones with Asterisk
Cisco IP Phone 7960 Administrator Guide for SIP, Version 1.0
Cisco IP Phone 7960 Administrator Guide for SIP, Version 2.0
SIP IP Telephone 7940/7960 Software

I got Windows server, Fletsphones and Cisco phones, and I got account on Voipbuster with free calls to Japan and other countries. Why can’t I make free calls from these phones? So here is my try at Asterisk.

1. Install Trixbox (formerly Asterisk@Home)
Since Trixbox is CentOS based, and I don’t have a free machine, I took a VMware image found here, (See "Introducing A Plug-And-Play Asterisk PBX for Windows") and run it on top of a Windows server via VWware Player. Note: set the VMware player to use NAT and DHCP.

2. Setup Asterisk to use Voipbuster account as outgoing trunk
Open freePBX interface of Trixbox, then click on "Trunks" menu. Make a little tweak to "SIP/voipdiscount" trunk just for quick test as follows:

PEER details:
allow=ulaw&alaw
authuser=asiboro
disallow=all
fromdomain=voipbuster.com
fromuser=<username>
host=sip.voipbuster.com
insecure=very
nat=yes
qualify=yes
secret=<password>
type=peer
username=<username>

Register string
<username>:<password>@sip.voipbuster.com

Adjust dialplan strings on both "Trunk" and "Outbound Routes".

3. Setup VP-1000
Setup VP-1000 to use Asterisk as SIP proxy/registrar. For example, use the pre-set number 500 (password 1234). Then VP-1000 can dial to whatever number allowed by dialplan via Voipbuster.

4. Setup Cisco IP Phone 7960
This is a bit not straightforward, since 7960 is shipped with SCCP ("Skinny" protocol) firmware by default, and this one is no exception. Though some people say that Asterisk can support Skinny protocol, it’s a lot easier if the firmware is replaced with SIP firmware. So I replaced SCCP to SIP and the rest is like setting any SIP user agent.