We've setup an Asterisk server with old Cisco 7911 IP phones. You'll find here the our installation and many configuration files.
More information about this phone: https://www.cisco.com/c/en/us/support/collaboration-endpoints/unified-ip-phone-7911g/model.html
Helpful Site for configuration: http://usecallmanager.nz/ and https://www.whizzy.org/2017/02/cisco-7941-asterisk-and-sip/
We use the Asterisk server for SIP and a tftp server that distributes the configuration to the phones.
apt-get install asterisk atftpd
Edit /etc/default/atftpd
USE_INETD=false
# OPTIONS below are used only with init script
OPTIONS="--tftpd-timeout 300 --port 69 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.239.239.0-255 --mcast-ttl 1 --maxthread 100 --verbose=5 /srv/tftp"
See asterisk/ folder for the configuration files, run ./makeConfig.sh in that folder and copy all .conf files to /etc/asterisk/.
The server runs as a kvm image.
virt-install \
-n spacephone \
--description "Asterisk" \
--os-type=Linux \
--ram=512 \
--vcpus=1 \
--disk path=....../spacephone.qcow2,bus=virtio \
--network bridge:br0 \
--accelerate \
--import \
--os-type=linux \
--os-variant=debianetch \
--noautoconsole \
--vnc
We used our dhcp server (ISC dhcpd) to give the sip phone the ip address of the tftp server.
Somewhre in /etc/dhcp/dhcpd.conf
class "cisco"
{
match if substring (option vendor-class-identifier,0,37) = "Cisco Systems, Inc. IP Phone CP-7911G";
# phone.mainframe.lan
option tftp-server-name "<IP Address HERE>";
}
Depending on your sip phone you have to change the match string.
We use lighttpd that serve only the www/phonebook.xml.
apt-get install lighttpd
Add to /etc/lighttpd/lighttpd.conf
mimetype.assign = ( ".xml" => "text/xml" )
Download the SIP firmware from here. Use the zip version and unzip the content into tftp/
Create and edit the secrets file.
cd phone
cp secrets.example.txt secrets.txt
./makeConfig.sh
copy the tftp/ folder to /srv/ftfp
Copy everything in www/ to /var/www/.
Use this credentials after ssh login: User/Pass: debug/debug
Reboot the phone with **#** (in the menu).
tcpdump -vvlenx -s 1500 -X port 5060 or port 5061
tcpdump -vvlenx -s 1500 -X ether host e8:40:40:a2:3b:ef
Open asterisk console:
asterisk -crvvvvv
Helpful commands
sip set debug on
sip reload
core reload