3.2.10. Cisco 7960 IP Phone - SIP firmware version
1.Introduction
2. Prerequisites
3. TFTP Configuration
1. Install the Solarwinds TFTP Server
2. You have to configure the server to transmit and receive files
4. Asterisk PBX configuration
For more information about how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk
5. Cisco configuration files
6. Final steps
7. Configurations from the phone menu
8. Additional information
The Cisco 7960 IP Phone is a hardphone which supports the Skinny Call Control Protocol(SCCP) to run with Cisco CallManager, the Session Initiation Protocol(SIP) and also the Media Gateway Control Protocol(MGCP). This is possible because it can load different firmware versions on bootup. This happens through a TFTP server. In this tutorial we will explain to you how to configure your phone for SIP and Asterisk PBX.
2. Prerequisites
Firstly you need the latest version of the SIP image file. In our case this is version 7.4. You can download it from here - http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960.
The .zip file should contain the following:
1) OS79XX.TXT - text file in which you should write the name of the image version (IMPORTANT This file is case sensitive and it must only contain the name of the file that you are attempting to load, without the .bin extension.)
2) P003-xx-y-zz.bin - where xx-y-zz is the number of the image version
3) P003-xx-y-zz.sbn - where xx-u-zz is the number of the image version
4) P0S3-xx-y-zz.loads - where xx-u-zz is the number of the image version and "S" is for SIP
5) P0S3-xx-y-zz.sb2 - where xx-u-zz is the number of the image version and "S" is for SIP
2) P003-xx-y-zz.bin - where xx-y-zz is the number of the image version
3) P003-xx-y-zz.sbn - where xx-u-zz is the number of the image version
4) P0S3-xx-y-zz.loads - where xx-u-zz is the number of the image version and "S" is for SIP
5) P0S3-xx-y-zz.sb2 - where xx-u-zz is the number of the image version and "S" is for SIP
You need five more files - SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml. You have to create them. If you don`t know what to write in this files - below I am giving you a .tar.gz file with an example of their contents. You have to change just a few things.
You also need a TFTP Server. We are using Solarwinds for Windows. You can download it from here http://solarwinds.net/
Of course you need and a working Asterisk PBX with registered users in sip.conf and made extensions.
Of course you need and a working Asterisk PBX with registered users in sip.conf and made extensions.
3. TFTP Configuration
1. Install the Solarwinds TFTP Server
2. You have to configure the server to transmit and receive files
4. Asterisk PBX configuration
Firstly you need to register an user in the sip.conf file. In our case we are registering the user pstn01 which we will use for line1 and user pstn02 for line2. You have to set also the password(secret) - in our case pstn01 for line1 and pstn02 for the second line.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf
This were the configurations in sip.conf
Now lets see the extensions.conf file. You can see a screenshot below. The first three lines of the context test show that if somebody dials cisco_line1 or the number 100 his call will be directed to pstn01 through SIP. In our case this is line 1. The next three lines show the same but for the second line - if somebody dilas cisco_line2 or 101 his call will be directed to pstn02 through SIP.
Now lets see the extensions.conf file. You can see a screenshot below. The first three lines of the context test show that if somebody dials cisco_line1 or the number 100 his call will be directed to pstn01 through SIP. In our case this is line 1. The next three lines show the same but for the second line - if somebody dilas cisco_line2 or 101 his call will be directed to pstn02 through SIP.
For more information about how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk
5. Cisco configuration files
1) With an editor such as Notepad for windows or vi for linux open the SIPDefault.cnf and add/write the image version you are going to use. You have also to write down the IP address of your Asterisk server. Its`s very important to set the proxy_register to 1. This will make the phone to register itself in your Asterisk PBX
Example:
; sip default configuration file
#Image Version
image_version:P0S3-07-4-00 ;
#Proxy server address
proxy1_address: 10.3.3.35 ;
proxy_register: 1;
#Image Version
image_version:P0S3-07-4-00 ;
#Proxy server address
proxy1_address: 10.3.3.35 ;
proxy_register: 1;
There are lots of parameters you may change here but this are the obligatory.
If you use NAT don`t forget to set nat_enable: to 1(0 -disable; 1- enable), nat_received_processing: to 1(0-disable; 1-enable) and to show the NAT IP address.
If you use NAT don`t forget to set nat_enable: to 1(0 -disable; 1- enable), nat_received_processing: to 1(0-disable; 1-enable) and to show the NAT IP address.
Example:
nat_enable: 1
nat_address: 10.10.0.1
nat_received_processing: 1
nat_address: 10.10.0.1
nat_received_processing: 1
2) In the second file (SIP<macaddress>.cnf) you have 3 parameters to change
a) linen_name - where n is the number of the line i.e. 1,2,3 and so on. This name could be whatever you want.
b) linen_authname - here you should to write the name of the user which you have already registered in your Asterisk PBX and you want to use for this line. For example the user pstn01.
c) linen_password - here you write the password(secret) which you have set for this user in your Asterisk PBX. In our case - pstn01
b) linen_authname - here you should to write the name of the user which you have already registered in your Asterisk PBX and you want to use for this line. For example the user pstn01.
c) linen_password - here you write the password(secret) which you have set for this user in your Asterisk PBX. In our case - pstn01
Example:
; phone-specific configuration file sample
line1_name : pstn01
line1_authname : pstn01
line1_password : pstn01
line1_name : pstn01
line1_authname : pstn01
line1_password : pstn01
In the name of this file you have to replace <macaddress> with this of your own phone. You can find it written on its back side in the bottom middle. For example - SIP00D29C01980.cnf (don`t forget that the MAC address should be with capital letters but .cnf it`s in lower case).
3) In xmlDefault.CNF.XML you fill the IP address of yur Asterisk PBX and you have to add this tag: <loadInformation7 model="IP Phone 7960">P003-07-4-00</loadInformation7>. Don`t forget that in our case the image version is 7.4 - that`s why in our tag we wrote P003-07-4-00. If you are going to use another version this numbers will be different.
4)RINGLIST.DAT - This is the file where you show the ringtone files.
For example:
Piano 1 Piano1.raw
Piano 2 Piano2.raw
Pop Pop.raw
Pulse Pulse1.raw
Old Style ringer1.pcm
Synth Low ringer2.pcm
Piano 2 Piano2.raw
Pop Pop.raw
Pulse Pulse1.raw
Old Style ringer1.pcm
Synth Low ringer2.pcm
5)dialplan.xml - In this file write the following:
For example:
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
6. Final steps
The next step is to unzip the files in the root directory of your TFTP server including SIPDefault.cnf, SIP<macaddress.cnf>, xmlDeafualt.CNF.XML, RINGLIST.DAT, dialplan.xml.
Reset your Cisco 7690 IP Phone by unpluging the power cord
Then if everything is ok you should see on your TFTP server something like this:
Then if everything is ok you should see on your TFTP server something like this:
Part from the log file:
5/08/2005 17:25 :Sending SIPDefault.cnf to (10.0.1.12)
5/08/2005 17:25 :Sent SIPDefault.cnf to (10.0.1.12), 1321 bytes
5/08/2005 17:25 :Sending SIP000D29C01980.cnf to (10.0.1.12)
5/08/2005 17:25 :Sent SIP000D29C01980.cnf to (10.0.1.12), 1129 bytes
5/08/2005 17:25 :Sending dialplan.xml to (10.0.1.12)
5/08/2005 17:25 :Sending RINGLIST.DAT to (10.0.1.12)
5/08/2005 17:25 :Sent dialplan.xml to (10.0.1.12), 109 bytes
5/08/2005 17:25 :Sent RINGLIST.DAT to (10.0.1.12), 0 bytes
5/08/2005 17:25 :Sent SIPDefault.cnf to (10.0.1.12), 1321 bytes
5/08/2005 17:25 :Sending SIP000D29C01980.cnf to (10.0.1.12)
5/08/2005 17:25 :Sent SIP000D29C01980.cnf to (10.0.1.12), 1129 bytes
5/08/2005 17:25 :Sending dialplan.xml to (10.0.1.12)
5/08/2005 17:25 :Sending RINGLIST.DAT to (10.0.1.12)
5/08/2005 17:25 :Sent dialplan.xml to (10.0.1.12), 109 bytes
5/08/2005 17:25 :Sent RINGLIST.DAT to (10.0.1.12), 0 bytes
7. Configurations from the phone menu
Now lets take a look at the manual configurations from the phone menu.
After the initiation press the settings button. Go to the last line called Unlock Config. Write down the password to unlock the phone. By default it is cisco.
Go back to line four - SIP Configuration. Enter the settings for line 1.
If you have set more lines in SIP<macaddress>.cnf do the same for them.
Your phone is ready to be used :)
After the initiation press the settings button. Go to the last line called Unlock Config. Write down the password to unlock the phone. By default it is cisco.
Go back to line four - SIP Configuration. Enter the settings for line 1.
1) Name - you can write here whatever you want this is not the name of the line but the name of the phone for example you can write SALES
2) Shortname - write the name you want to see for the line. For example Line 1
3) Authentication Name - write te name of the registered user in your Asterisk PBX. In our example pstn01
4) Authentication Password - the secret for this user whish is set in the Asterisk PBX. In our example pstn01
5) Proxy Server - the IP address of your Asterisk. In our example 10.3.3.35
6) Proxy Port - by default it is 5060. Keep it the same.
2) Shortname - write the name you want to see for the line. For example Line 1
3) Authentication Name - write te name of the registered user in your Asterisk PBX. In our example pstn01
4) Authentication Password - the secret for this user whish is set in the Asterisk PBX. In our example pstn01
5) Proxy Server - the IP address of your Asterisk. In our example 10.3.3.35
6) Proxy Port - by default it is 5060. Keep it the same.
If you have set more lines in SIP<macaddress>.cnf do the same for them.
Your phone is ready to be used :)
8. Additional information
If you want to know how to configure your Cisco 7960 IP Phone to work with the skinny protocol (SCCP) and Asterisk PBX just click here
| Add Comment |
Latest Headlines:
-
T.38 faxing with Zoiper 2.15 is now easier than ever
section: voip software
-
Asterisk 1.4.21 Released
section: Asterisk
-
Asterisk 1.4.20 Released
section: Asterisk
-
Asterisk 1.4.20-rc2 Released
section: Asterisk
-
Asterisk 1.4.20-rc1 Now Available
section: Asterisk
- News Archives (older news)
Latest Tutorials:
-
Sending Fax from Zoiper to Zoiper using T.38
added 08/Dec/2008 18:16
-
VMAuthenticate (dialplan application)
added 01/Mar/2008 15:57
-
Siptronic ST-530
added 06/Nov/2007 17:57
-
Siemens C455 IP hardphone
added 05/Nov/2007 10:24
-
Zoiper
added 22/Oct/2007 17:53
Latest Comments:
-
Здравствуйте
Приятн...
tutorial: SIP with NAT or Firewalls
-
https://muckrack.com/sarah-viloid/bio
h...
tutorial: General Installation
-
https://megadonald.taplink.id
https://a...
tutorial: Background (dialplan application)
-
https://www.utgop.org/missionimpossiblef...
tutorial: asterisk.conf
-
Alpha Grow Male Enhancement :- products ...
tutorial: Basic Information