Cisco 7960 IP Phone - SIP firmware version

3.2.10. Cisco 7960 IP Phone - SIP firmware version

1.Introduction

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

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.

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

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.

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;

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.

Example:

nat_enable: 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

Example:

; phone-specific configuration file sample
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

5)dialplan.xml - In this file write the following:

For example:

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:

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

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.

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.

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

User Comments

mahmodi (mostafamahmodi at yahoo dot com)
18 November 2008 08:31:09
please send IOS cisco ip phone 7960 , ios file name P00307020300.bin
best regard

Mauricio (mauricio dot ramirez at lacity dot org)
25 September 2008 21:57:03
Anyone know if it's possible to set up this phone or any Cisco phone to Auto-Dial when it goes off-hook. Like the bat phone in the old Batman shows?

Syed Ahmed (dewyplatinum at gmail dot com)
21 September 2008 15:49:28
Fine. Please send me update news from your company. Thanks

mohammad (malmno_19 at yahoo dot com)
10 September 2008 21:36:28
key of cisco 7960 general
verision 2000-2003

Ashish (ashash210 at hotmail dot com)
05 September 2008 10:26:12
hi, i am running my call manager on a vmware i have installed it on a server and given it a different ip address but on the same vlan network i can reach the call manager from a different pc sitting on the same network but the ip phones keep giving me TFTP timeout that's on the 7940 the rest keep saying configuring ip yet all the phones and the server are on the same vlan and same network can you guide me please thanks

Yiannis (eastern at pathfinder dot gr)
12 August 2008 19:51:44
ONLY an IDIOT would talk about a link to a tar.gz file and then would not give it out.

francis (francisdtan at gmail dot com)
13 June 2008 06:33:03
I have a cisco 7960, when I hook it up on my dsl, it hangs in configuring CM list ( must be a callmanager thing in a server) whew! can I use my cisco the way you do, meaning using SIP? how to I connect the phone? what I did, i connected my dsl to the phone
then from the phone to my PC and installed sip,
what do I need to make the phone make calls?

ray (iconnectoid at gmail dot com)
06 June 2008 20:49:25
Great tuturial. Best one of I have read so far.

My trixbox tftp serves CTLSEP000F8F4E4E5B.tlv repeatedly and the phone resets time after time. I have seen no documentation on this behavior nor can I find this 'CTLSEP000F8F4E4E5B.tlv' in the Centos filesystem or reference to it online.

My tftp is complete? The endpoint mgr is configged (not that it matters at this stage.)

Any thoughts or good jokes would be huge.

Doona Pierro (dpierro at caravaningredients dot com)
06 June 2008 15:37:14
I have a question, While listening to my messages I accidently deleted a message, any way of getting it back??

waleeded (ws_forever25 at yahoo dot com)
06 May 2008 11:11:20
pleas if you can help me
i have call manger in my campany and our problim is whene we connect the the internet in the computer tru the ip phone it will be so slow, but if we will connect the computer in the internet with out the ip phone the internet we be faster.
if you have see like this problim pleas try to lett me know abut the solution
thank you

sun (sunwa36 at yahoo dot com)
20 April 2008 13:18:27
Can you help me?I need five more files - SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml.
For Cisco 7960 IP Phone

vu trong hieu (hieu_vutrong at yahoo dot com)
18 March 2008 11:41:37
I got some Cisco 7760 with SIP version 8.2 from my predecessor without any CDs. Do I have to pay Cisco to download the files SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml ??? I don't have CCO account. Would anyone plz send me these files :-) thanks in advance. cheers.

John Doe (heppy7000uk at yahoo dot co dot uk)
12 March 2008 00:07:09
Please Please help!!

In the tutorial its says:

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.

How do I create these files?? Are they just text files with the .cnf extension? Alternatively if anyone has these files could they ping them over. I have the phone communicating with the tftp server and its looking for the files but they dont exist.

They dont make these things easy do they!!

nny (info at atlantiatech dot com)
26 February 2008 23:47:07
i am gonna have to agree, no tar.gz example = epic fail!

Ilan (ilanh at excite dot com)
05 January 2008 10:05:27
An explanation for the content of tar.gz files can be found at:
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/1_0/english/administration/guide/install.html

In addition to manual configuration and unlocking info.

Florida Guy (unknown at unknown dot com)
21 December 2007 21:06:14
##*## Resets the phone, takes about two minutes

Iftikhar (iftikhar at infotex dot biz)
23 November 2007 09:01:56
Zdravstvuyte!
Menya interesuyet, mojno li proshit cisco ip-phone 7931 v SIP versiyu, eslida, to ne podskajite gde eshe proshivku na nego dostat'?
Zaranee spasibo.

Greg Cragg (bedrock27 at hotmail dot com)
08 November 2007 18:25:44
Was reviewing your tutorial for the upgrade for the 7960 to Sip. I downloaded the requisite files from Cisco but can't seem to find the additional five required files you have listed: SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml. I believe you stated they were listed below in a .tar.gz file but I found no such file listing. If you could point me in the right direction that would be greatly apppreciated. Thanks.

Dave (dgagne at netfast dot com)
29 October 2007 16:12:50
Article on SIP v. Skinny......

asdads (asd at asd dot asd)
27 September 2007 17:23:00
This is worthless, useless and a waste of all of our time without the example file.

Jordi (jguiu at bauwerk dot es)
13 September 2007 13:58:47
Can you send me a mail with a .tar.gz with the configuration files??

Thanks!!

mark (markdavid at caramail dot com)
29 August 2007 15:30:37
have setup everything like you said above I am running TRIXBOX with cisco 7960 phone. but i ignore how make addresse with DHCP and where i can make a DHCP SERVER

Ray (raydeluca at gmail dot com)
13 July 2007 19:40:38
I have setup everything like you said above I am running asterisknow beta with cisco 7960 phone. I can reach everything in asterisk voicemail, prompts you name it .... But i can not reach extensions or dial out. ??????? Please help i have been looking over this for weeks

Santiago (sanriv at gmail dot com)
13 June 2007 23:25:42
Can anyone know how to block all buttons of the cisco 7940?, I need this because I have connected my 7940 to SIP server Asterisk. the inbound calls are received through softphone(made by company) on computer. but I want that buttons does not working while I use the cisco 7940. I refer, to disable menu button, settings button, all keypad, etc. Everything it is done through softphone. Can anyone help me?

Thomas (t dot roemhild at arcor dot de)
27 May 2007 08:53:47
Hi,
does anyone know where can i get SIP
Image P0S3-04-0-00, I need this one to upgrade my 7960 in Steps.

Thx
Cheers
Thomas

Carlos (CARLOSV72 at HOTMAIL dot COM)
17 May 2007 08:25:12
This is the Sipdefault.cnf and sipxxxxxxxxxx.cnf I'm using:
SIPDEFAULT.CNF:
; sip default configuration file
# Image Version
image_version: "P0S3-08-2-00"
# Proxy Server USE YOUR SIP SERVER ADDRESS OR DOMAIN NAME
proxy1_address: "xx.xx.xx.xx"
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "180"
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "g729"
# TOS bits in media stream [0-5] (Default - 5)
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"
# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: "avt"
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: "3"
# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "10" ; Default 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
telnet_level: "2"
nat_enable: "1"
#nat address I put the public IP address at the site where the phone is, but I don't really think it makes a difference
#nat_address: "XX.XX.XX.XX"
nat_received_processing:"1"
# Setting for Message speeddial to Voicemail
#messages_uri: "9195551000"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "216.152.68.16"
#sntp_server: "sntp.company.com"
time_zone: "EST"
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 101
# XML file that specifies the dialplan desired
dial_template: "dialplan"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"
#Enable or Disbale VAD (0-disabled (default), 1-enabled)
enable_vad: "0"
phone_password: "cisco"
;services_url: "http://www.company.com/phone/services.asp"
;directory_url: "http://www.company.com/phone/companydirectory.asp"
# here yoou put a link to a logo in a webserver, it will show on the screen.
logo_url: "http://www.logvault.net/logo.bmp"

SIPXXXXXXXXX.CNF:
; phone-specific configuration file sample
line1_displayname: "5555820031"
line1_name: "5555820031"
line1_shortname: "5555820031"
line1_authname: "5555820031"
line1_password: "XXXX"
line2_displayname: ""
line2_name: ""
line3_displayname: ""
line3_name: ""
line4_displayname: ""
line4_name: ""
line5_displayname: ""
line5_name: ""
line6_displayname: ""
line6_name: ""
nat_enable: "1"
nat_address: "XX.XX.XX.XX"
nat_received_processing: "1"
phone_label: "MY PHONE"
phone_prompt: "THIS IS THE TELNET PROMPT"
callerid_blocking: "0"
dtmf_outofband: "avt"
network_media_type: "auto"
dtmf_avt_payload: "101"
call_waiting: "1"
cnf_join_enable : "1"
semi_attended_transfer : "1"

Paul (paul at adaline dot nl)
09 May 2007 23:16:30
Does anybody know how to set the registration interval on the phone or * PBX ?

Ben (nuage29 at yahoo dot com)
03 May 2007 22:15:10
HI,

Could you please email me the tar.gz file that you reference in the article?

Thanks

Shon (shon at misproductions dot com)
27 April 2007 01:54:35
I put everything in as necessary, the phone reads the OS79XX.TXT file, loads the xmlDefault.cnf.XML file, but does not load the new firmware which was the only firmware available from Cisco which was POS3-08-2-00. The phone currently has Cisco SCCP software, P0030301MFG2 (3.10MF.G2)

can anyone please help me out?

moya (moya at korangan dot com)
24 April 2007 08:29:38
no korangans

prasad (prasad dot hv at sylantro dot com)
03 April 2007 11:50:36
please tell me how to find the previous firmware version does the 7960 series CISCO IP phone has? it showing "protocol application Invalid".

Chris James (cjames at powerassociates dot com)
26 March 2007 21:19:52
Can some please send me the tar.gz file he is talking about? I cannot fine it anywhere.

Thanks, Chris

Giuseppe (djrichard74 at hotmail dot com)
15 March 2007 12:39:40
Hi,

How I do reset the phone?
Thank you for the help

Regards

muse (darkness_less at hotmail dot fr)
03 March 2007 12:40:20
Hi,
I am trying to find the example files for the followingSIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml. etc.
I cannot find the tar.gz file that you refer to.

Andres (Andres at europesip dot com)
06 February 2007 15:47:08
Hi, I upgraded the firmware to release P003-08-6-00, and I althought the SIP uggrade went fine, and the phones can now do calls, I have a problem with the registration on asterisk. Problems are that CISCO 7940's aren't showing in my "sip show peers", however i can make out going calls no problem , but obviously incoming calls are a problem . all my other brand handsets , Grand streams / Sipura / Xten lite, don't have this problem. sip.conf config's for both all extensions are the same .

I need an urgent solution for this problem, I would appreciate any info or suggestion. My contact is andres (at) europesip (dot) com

Lijet (lijettranspress at superig dot com dot br)
02 February 2007 16:02:38
segue anexo

voip phone (voiptop at 126 dot com)
31 January 2007 01:44:58
deleted - [spam]

Safir (msafirs at gmail dot com)
26 January 2007 14:51:41
Brian, Contact me I will provide you with the version 6.3.

Brian (brianrayner01 at tiscali dot co dot uk)
17 January 2007 18:28:08
I require version 6.3 to load before going to 8.2, where do I get the software from?
I will also need the example .tar.gz files.
Do I need to load 7.3 before 8.2?
Many thanks, Brian.

hello (dont_have at one dot com)
17 January 2007 17:49:21
all you need to go from 6 non-SIP to 8-SIP firmware:
http://www.redbrick.dcu.ie/~norim/soft/7960_firmware_form_6_to_8.rar

kyriakos (kyriakos at otenet-telecom dot com)
10 January 2007 10:59:49
Hi, can i please have the sample configuration files for sip?

vanch (vanhut84 at yahoo dot com)
28 September 2006 09:07:12
can u send me tarball

Hunter (battleserver at yahoo dot com)
26 September 2006 02:28:25
If your in Asia or Euro, and cant get your 7960 to work, or if your stuck with the Universal Application Loader. There is a company who I had sent my phone to and for 50$ plus shipping, I paid a total of 85$ but had my phone in a few days. Just email them for a "RON" Number (Repair Order Number) and ship your phone. If you ship no need to ship your power cube. Now my phone is running on my asterisk with no glitch. Your can email the company direct at spookdude@hotmail.com

radjesh chilar (r dot chilar at connectdatasolutions dot nl)
19 September 2006 13:16:41
.

gerson morales (guitarra at peru dot com)
18 September 2006 23:26:54
Good morning dear friends: Please, Can you send me this course os asterisk to my e-mail: guitarra@peru.com. I´m peruvian, also I´m electronic student. Thanks.

Rashid (rashid_info at yahoo dot com)
16 September 2006 17:59:08
Dear friend please help in this regard
One IP phone Problem

Model: Cisco 7960 Series
Problem:Versin Error coming , IP , SUBNET MASK, ROUTER ,TFTP SERVER, CALL MANAGER Set properlly But the problem coming is Versin Error and Registration failed.

Please help fast.

Rgds
rashid

Rashid (rashid_info at yahoo dot com)
16 September 2006 17:57:52
Dear friend pleax help in this regard

Problem:Versin Error coming , IP , SUBNET MASK, ROUTER ,TFTP SERVER, CALL MANAGER Set properlly But the problem coming is Versin Error and Registration failed.

Please help fast.

Rgds
rashid

zoa (support at asteriskguru dot com)
01 September 2006 13:49:42
You have to license it from cisco. (you can get it from their website.)

We have no rights to put it online.

Adrian (abarrios at iplan dot com dot ar)
23 August 2006 15:22:38
Hi,

I am trying to find the example files for the SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml.
I cannot find the link to the tar.gz file.

Thanks in advance.

LJR (msn20 at hotmail dot com)
13 August 2006 02:36:43
Can Anyone help me out? My cisco 7960 has the Universal Application Loader (version unknown) and I cant get it to do anything. I tried looking at a number of sites, which tell me to use the SIPDefault.cnf and the XML files, however, it doesnt do anything, other then request the information. I assume that that the Application loader is version 7++ but from my understanding the version 6 was the most stable. I do have all the firmware, but I cant get the phone to reload the version 6. When I look at the tftp, it doesnt even request for the OS79xx file, so thats off my list. since the Application loader is differ, I was under the impression that there is a different command or file you can config so the phone can load the new firmware. I dont have CallManager, so would not know what to do. Any advise??

Robq (rob_burcham_NO_SPAM at yahoo dot com)
08 August 2006 01:16:48
Take out the _NO_SPAM to email me...

I have upgraded to P0S3-08-2-00 (very difficult and confusing) from a skinny load. Now it is sip, and it reads the SIPDefault and processes all the directives, like NTP server and logo image URL, and all these things work. However the status shows:

W310 1 Error(s) Parsing: SIPDefault.cnf

And the phone never even tries to register to my asterisk box. The asterisk console never even shows an attempt. I have other 7960s running P0S3-06-3-00 and they do register, and they are using the same configs.

You cannot downgrade the 79xx phones. That's right - you CAN'T. If I try to point a skinny phone or a P0S3-08-2-00 phone to my P0S3-06-3-00 image, the phone reports "Version Error" and fails.

Does anyone have a P0S3-08-2-00 working? Care to share your SIPDefault.cnf and SIP<mac>.cnf? Please?

luke (luke at computergurus dot net)
01 August 2006 00:27:41
Do you have to go from 6.3 to 7.3 before loading 8.2? If so, I can only find 8.2 on cisco's site. Any idea where I can find 7.3?

Robert (r at ocg dot nu)
27 July 2006 20:19:03
Many SIP Firmware versions, including POS3-07-5-00 cannot handle a double quote when passed in the Asterisk variable Caller ID Name CALLERID(name) string.

Dave Rose (drose at godaddy dot com)
28 June 2006 23:18:57
has anyone got a link to the configuration file tar ball?

Dave

Greg Beason (greg at gregbeason dot com)
21 June 2006 08:42:48
I cannot get my Cisco 7960 (firmware 7.2) to process nat properly. It will connect to my asterisk server that is not behind any nat fine for the first 5 minutes, but after that, it will not allow any dialouts. All dialouts will simply say Calling (out INV). On the asterisk server, it displays the status of the device as unreachable. I have to reset the equipment and then I can access the server. Does anyone have any suggestions?

SimonD (simon dot darlington at det dot nsw dot edu dot au)
24 May 2006 07:54:37
Hi,
I am trying to find the example files for the followingSIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml. etc.
I cannot find the tar.gz file that you refer to.

Nitin (nitintreddy at yahoo dot co dot uk)
12 May 2006 22:10:37
SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml.
I am trying to download your .tar.gz file i dont find it in your form can you pls give me a link for it

Nitin (nitintreddy at yahoo dot co dot uk)
12 May 2006 21:57:50
SIPDefault.cnf, SIP<macaddress>.cnf, xmlDefault.CNF.XML, RINGLIST.DAT and dialplan.xml.
I am trying to download your .tar.gz file i dont find it in your form can you pls give me a link for it

Nick (asteriskguru dot com at bustin dot co dot uk)
29 April 2006 18:15:29
Anyone know where I can source the SIP 6.3 firmware? I cannot find it anywhere on the Cisco website.

Ryo (matlego at caramail dot fr)
27 April 2006 16:57:50
This tutorial works for cisco7912G ?? Thks

Luis (sousaluis at yahoo dot com)
06 April 2006 15:18:23
Hi, how can I find the older versions of the firmware. I'm only able to find at cisco the latest version the 8.2.And I really need the 6.3 because my current firmware is 3.3.

Thanks

sigmounte (asteriskguru at sighq dot net)
27 February 2006 02:36:08
how the ip phone know the address of the tftp server ??

do i need to tell it in my dhcpd.conf ?

Jaison (jaittu at gmail dot com)
13 February 2006 16:11:53
It works fine with me. Today I configured two 7960s. Thanks

herath (prkherathelearning at yahoo dot com)
02 February 2006 04:36:51
I am using Micronet SIP phone. But the documentation available does not cover it. Please add an documentation clearly describing hoto configure an Micronet SIP phone using Asterisk

Yhank you

herath (prkherathelearning at yahoo dot com)
02 February 2006 04:36:38
I am using Micronet SIP phone. But the documentation available does not cover it. Please add an documentation clearly describing hoto configure an Micronet SIP phone using Asterisk

Yhank you

herath (prkherathelearning at yahoo dot com)
02 February 2006 04:36:21
I am using Micronet SIP phone. But the documentation available does not cover it. Please add an documentation clearly describing hoto configure an Micronet SIP phone using Asterisk

Yhank you

krish (krishiin at gmail dot com)
03 January 2006 16:42:58
hi, i am krish,i have one doubt in configuration of cisco SIP phones.we have configured one cisco sip phones through our internet gateway server.we have assigned local ip address for that sip phone,and gateway is our internet gateway and also enabled NAT.In the internet gateway server we have configured NAT also.now ,the whole setup is working fine.but i cannt hear any voice any from calling device.which is located in the out of our network.but i can hear by press HOLD and RESUME button after reciving calls.what should br the problem.please help me out of this...

Thanks in advance

krish

krish (krishiin at gmail dot com)
03 January 2006 16:41:34
hi, i am krish,i have one doubt in configuration of cisco SIP phones.we have configured one cisco sip phones through our internet gateway server.we have assigned local ip address for that sip phone,and gateway is our internet gateway and also enabled NAT.now ,the whole setup is working fine.but i cannt hear any voice any from calling device.which is located in the out of our network.but i can hear by press HOLD and RESUME button after reciving calls.what should br the problem.please help me out of this...

Thanks in advance

krish

Carl Johansson (mail at carl-johansson dot com)
26 November 2005 09:57:42
Hi,

I understand most of this but I have not been able to download any of these files from Cisco. Does anyone know where to find them?

Adam (adam at solwinder dot com)
20 November 2005 07:57:02
Hi, I dont see the tar.gz. Please include it thanks.

day (dayhox at gmail dot com)
01 September 2005 23:10:08
CISCO CSE email Re: SIP IP Phone Upgrade

*** NOTE: Please read thru these directions completely before you begin ***

You will typically need be concerned with only 3 files, OS79XX.TXT,
SIPDefault.cnf, and *.bin. You must have these 3 at least in the root
directory of your tftp server (there may be more and that is ok, but you
must have these 3). You can download each of these from CCO.

If you are not already up to SIP 6.3 or above or if you are running a load
other than SIP, you must first upgrade to SIP 6.3, then to 7.3.

To upgrade to 6.3,

1) make sure that the OS79XX.TXT contains only the filename P0S3-06-3-00
(these digits are all alphanumeric zeros, not letters)

2) make sure that the SIPDefault.cnf contains P0S3-06-3-00 in the
image_version field

3)make sure phone has the correct ip address of the TFTP server

4)remove/reattach power from phone for it to go through the correct upgrade
procedure (do not use reset sequence). The phone will powercycle itself after
a successful upgrade).

To upgrade to 7.3

1) change the OS79XX.TXT to P003-07-3-00 (Note: this filename is now different
than the filename in SIPDefault.cnf - see step 2)

2) change the image_version in the SIPDefault.cnf file to P0S3-07-3-00 (please
note the filenames in the OS79XX.TXT and SIPDefault.cnf are 'different' during
for the upgrade from 6.3 to 7.3 only)

3) Follow steps 3 & 4 above

If you are having problems with your tftp server, try using Solarwind. If you have
problems with the tftp, you may want to refer to the tftp logs (Solarwinds logs are
easy to read).

You can use the SIPmacaddress.cnf file to configure the phone specific parameters
after the upgrade is complete but it is not needed during upgrade.

Note: During the upgrade from 6.3 to 7.3, it is perfectly normal for the phone to
ask for files you do not have during the final reboot and should not affect the
outcome of the upgrade.

**********************

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