1120 1140 configuration files

When an 1120 1140 phone boots up fresh, its going to need configurations.  We can manually type many of these configs into the phone, but I'm not covering that at this time.  Its so time consuming that I will focus on TFTP deployments.

When a SIP series 1100 or 1200 series phone boots, it first looks for the file with a name specific to its model name.  1120eSIP.cfg 1140eSIP.cfg 1230SIP.cfg

In my experience the information contained in these configuration files are the same, but I will focus on the 1120/1140 phones specifically.

Every phone that boots will attempt to access this file on the TFTP server that you've either assigned through DHCP scope or manually typed into the PROVISION section of the phone.

[LICENSING]
If your enviornment has a licence server to handle the tokens for avaya products, you can enable this.  For the most part the phones will work just fine.  Its using features of the phone itself that dont always work.  For example, USB Headsets are not available without a licence.

[FW]
This is the firmware you will use.  In this case its showing version 26.00 of the SIP firmware which was the version available at time of writing.
Download_mode AUTO means the phone will compare and if the version is the same or newer, it wont do anything.  If you change that to FORCED, everytime the phone boots up it will attempt to download and install that version...even if it already has it.  So leave on AUTO and just change to forced if you are trying to change firmware versions.

[USER_CONFIG]
I've always left this on AUTO.  I'll increment the VERSION number if I want the phone to reload the phone configuration MAC file (covered a little later) to change some configurations.

[DIALING_PLAN]
This points the phone to the location of the dialing plan for the phone itself.  This file would have number sequences that, should you dial and it matches, the phone will automatically dial it.  If no match, you will need to press "SEND" button on the phone.
Changes you make to this file would also require the VERSION number in that section to be incremented.

If you dont already have a file in your TFTP, create one appropriate to your phone and put in the data between the ############### symbols.

11XXeSIP.cfg:

############################################################
[LICENSING]
#un-comment next 3 lines for environments with Nortel/Avaya licence key servers. 
#DOWNLOAD_MODE Auto
#VERSION 000003
#FILENAME ipctoken.rev2.cfg

[FW]
#SIP FIRMWARE
DOWNLOAD_MODE AUTO
VERSION SIP1120e04.04.26.00.bin
FILENAME SIP1120e04.04.26.00.bin

PROTOCOL TFTP
SERVER_IP X.X.X.X
SECURITY_MODE 0

[USER_CONFIG]
DOWNLOAD_MODE auto
VERSION 0001
PROTOCOL TFTP

[DIALING_PLAN]
DOWNLOAD_MODE AUTO
VERSION 0001
PROTOCOL TFTP
FILENAME dialplan.txt
############################################################

Put the above file in your TFTP directory.  The phone will look for this.  Change SERVER_IP X.X.X.X to be the IP address of your TFTP server.


Save that file.

In asterisk (via FREEPBX GUI) create a SIP extension and save /apply those changes
Note the extension number and password.

Now you will need to create a file in the TFTP directory that contains a file specific to your phone.
So, note your MAC address of your phone, then create a file in the format:  " sipMAC.cfg "
So if your MAC was 00AF34FE your file would be " sip00AF34FE.cfg "

In this file substitute the following values.
XXXX = extension number
YYYYYY = PASSWORD
ZZZZZZZZ = IP of your asterisk PBX

##################################################################
AUTOLOGIN_ID_KEY01 XXXX@company.net
AUTOLOGIN_AUTHID_KEY01 XXXX
AUTOLOGIN_PASSWD_KEY01 YYYYYYYYYY

SIP_DOMAIN1 company.net

SERVER_IP1_1 ZZZZZZZZZ
SERVER_PORT1_1 5060

SERVER_RETRIES 1 1
DNS_DOMAIN company.net

SIP_PING YES
USE_RPORT YES
RTP_MIN_PORT 10000
RTP_MAX_PORT 20000
SIP_UDP_PORT 5060

AUTH_METHOD AUTH_INT
AUTOLOGIN_ENABLE USE_AUTOLOGIN_ID
PROMPT_AUTHNAME_ENABLE NO

VMAIL *98
VMAIL_DELAY 300
MAX_APPEARANCES 3
DEF_LANG English
DEF_AUDIO_QUALITY High
ENABLE_LLDP YES
ADMIN_PASSWORD 26567*738
ADMIN_PASSWORD_EXPIRY 0

DEF_AUDIO_QUALITY High
MAX_LOGINS 1
USB_HEADSET LOCK
ENABLE_USB_PORT YES
USB_HEADSET YES
EXP_MODULE_ENABLE NO
ENABLE_SERVICE_PACKAGE NO
IM_MODE DISABLED
AVAYA_AUTOMATIC_QoS NO
VQMON_PUBLISH NO
SIP_TLS_PORT 0
ENABLE_BT NO

#####################################################

Save this file once you've made the appropriate extension, IP and password changes.

Now, power cycle your phone!

It should register

Nortel Avaya 1100 series 1120 1140 1230 dialplans

Although all my 1100/1200 series phones are connected to asterisk, it shouldn't matter a whole lot.
SIP or UNISTIM connected to a different PBX

Assumptions are you are deploying your phones via TFTP already and are somewhat familiar with the update process of one of these models of phones.

Once the phones have the dialplan, if you off-hook dial, the phone will nearly instantly dial the number when you are done typing it.  No "SEND" button required in matching dialplan cases.

As of writing this i haven't worked out fully the trickery for international calls since the numbers are all different lengths, its a work in progress.

SO first, in your 11xxeSIP.cfg or 12xxSIP.cfg file have this entry in there.

[DIALING_PLAN]
DOWNLOAD_MODE AUTO
VERSION 0001
PROTOCOL tftp
FILENAME dialplan.txt

Save that.

Now create a new file in your TFTP directory called "dialplan.txt"  (or change to whatever you want, just make sure it matches in the .cfg file FILENAME section.

In dialplan.txt put in the following and save it.

/* ------------------------------------------------------------------- */
/* */
/* Avaya 1100-1200 series IP Deskphone Dial Plan */
/* */
/* ------------------------------------------------------------------- */
/* Domain used in the dialed URL of the SIP INVITE message */
$n="avaya.com"
$t=300
%%
/* DIGITMAP: 10 digits starting with 1 */
(1[2-9]x{9})|(1[2-9]x{9})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 10 digits starting with 91 */
(91[2-9]x{9})|(1[2-9]x{9})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 4 Digit Extensions starting with 88XX */
(88x{2})|(88x{2})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 4 Digit Extensions starting with 5xXX */
(5x{3})|(5x{3})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 7 Digit Extensions starting with 7digitdial */
([2-9]x{6})|([2-9]x{6})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: VOICEMAIL */
(*99)|(*99)# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 10 digits starting with 555 */
(555[2-9]x{6})|(555[2-9]x{6})# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 911 Emergency */
(911|911)# && sip:$$@$n;user=phone && t=300

/* DIGITMAP: 9911 Emergency */
(9911|9911)# && sip:$$@$n;user=phone && t=300

/* End of Dial Plan */


Now if you are newly deploying this feature, you'll need to update the 11XXeSIP.cfg file in your system by increment the "VERSION" under [USER_CONFIG].

Now in your phone press the "SERVICE" button, or it might look like a picture of a globe with two arrows pointing left and right.

When you press that, the menu on the phone will change and you should see "CHECK FOR UPDATES".  Pressing this will cause your phone to contact the TFTP and check its config for a change.  If it see the VERSION number in [USER_CONFIG] has increased, it will reload the entire file...which will include the dialplan.txt file.  in MOST cases your phone won't fully reboot, but some phones do for some reason.  During my testing and deployment, especially after the phones have a dialplan, they will easily load any future diaplan changes if you increase the "VERSION" under [DIALING_PLAN}

I've included some examples in the above configs that may apply to you with some easy modifications.