Cisco IP Phone Configuration with Asterisk

Getting the Cisco IP Phone 7970 G to work together with the software PBX Asterisk was something I had my hands on a couple of years back. Here’s how you can get them talking together.

You need a couple of things to get this working:

A functioning DHCP serverA functioning TFTP serverSIP Firmware from Cisco This is just a gzipped and tar’ed file.A functioning asterisk serverA Cisco IP Phone

According to a recent installation, the TFTP server must contain the following files

apps70.1-1-2-26.sbn
cnu70.3-1-2-26.sbn
cvm70sip.8-0-2-25.sbn
dsp70.1-1-2-26.sbn
jar70sip.8-0-2-25.sbn
SIP70.8-0-3S.loads
term70.default.loads
term71.default.loads
SEP.cnf.xml

The file you should pay the most attention to is the SEP.cnf.XML file, this is the configuration file. The configuration file is in XML format. You can find a sample configuration here that should work.

true
SIP




M/D/Y
Greenwich Standard Time


true



ccm-beta-5-1
CallManager 5.0 Beta Pub – 5.0.1.032

2000
5060
5061

2427
2428


ccm-beta-5-1





Disable
Disable
false

2000

2000

2000
IP ADDRESS TO SIP SERVER
5060

5060

5060
false

-1
Default
Default
120



USECALLMANAGER
5060
USECALLMANAGER
5060
USECALLMANAGER
5060
true


true
x-cisco-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
false
2
true
true
2
2
0
true


6
10
180
3600
5
120
120
5
500
4000
70
true
None

1
false
true
false
false
none
101
3
avt
false
false
3

2
false
15000
10
true
16384
32766


9

USECALLMANAGER
5060
3302
3302

2

3

false
3

4
5
7b452e87-4496-4762-e11f-b26751a1884b

true
false
false
true



5060
184
0

SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml



true
2

SIP70.8-0-3S

falsefalse01000001,707:3010:3001:001
1136931633-57191cee-5ffc-4342-b286-4246b4991890

English_United_States
1
en_US
1.0.0.0-1
iso-8859-1

United_States

United_States
64
1.0.0.0-1

1
0
http://ccm-beta-5-1:8080/ccmcip/authenticate.jsp
http://ccm-beta-5-1:8080/ccmcip/xmldirectory.jsp

http://ccm-beta-5-1:8080/ccmcip/GetTelecasterHelpText.jsp


http://10.86.5.102/CiscoServices/index.xml
96
0
96
4
0


3804
ccm-beta-5-1



false

2
2000
2000

true
true

dialplan.xml

On the Asterisk server, you will have a file named sip.conf and to have the Cisco IP Phone talking to Asterisk you need this

[999999999]
username=999999999
type=friend
secret=password
nat=no
host=dynamic
canreinvite=no
dtmfmode=rfc2833
context=incoming
qualify=yes
disallow=all
allow=ulaw

That should be it, good luck!

Tags: asterisk, cisco 7960, cisco 7970, cisco 7970G, cisco config, cisco ip phone, config files, configuration, ip phone

This entry was posted on Wednesday, May 20th, 2009 at 12:20 am and is filed under Cisco configuration, Tutorials. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.

OpenVox Announces Free Support for Asterisk Configuration

OpenVox has announced today to provide free support for asterisk configuration to the open source community. It can surely lower the technical entrance to the open source telephony for the new users.

OpenVox is based on open source, and we are working with asterisk, Askozia, Elastix, FreePBX, FreeSWITCH, PBX in a flash, trixbox… You’re always not limited by your choice of application. But some of you may step back when you are new to open source or don’t know what to do with it. And now you can get the real free technical support to make your telephony system work. There will no longer be any worries about the complex system configurations and settings.

What Do We Have for Free Support?
If you are installing OpenVox hardware with the following software:
Asterisk, Askozia, Elastix, FreePBX, PBX in a flash, trixbox.
Then, you can have the following free support from OpenVox:
*Product Driver Installation & Configuration
*Setting for Inbound/Outbound Routing
*SIP/IAX Extension Configuration (A few extensions to ensure the system working properly)

More will be added to the list soon. If you have any suggestions for the list or something else you think will be good, please don’t hesitate to email us at james.zhu@openvox.cn.

What Kind of Info Needed for the Support?
To access the free support, we will require you to provide the following information to us:

*Name of the Product
*Serial Number for the Product
*The Name of the Software and its Version
*SSH Account and GUI Login Info
*What kind of free support do you need from us?
*When will be your best available time for us to get it done for you? Please kindly note that our working hours is from 9:00 -18:00 (GMT +8) even though we can make it as flexible as possible to fit yours.

Please email the above information to James Zhu at james.zhu@openvox.cn.

When Will You Have the Support?
We will get back to you within one or two business days once your email is received. And the exact time will be confirmed with you afterwards. Well, enjoy the open source telephony!