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.

Use asterisk Direct Dial from IVR to goto a different PBX system

So lets say you are running a Cisco or other type of SIP capable PBX in your business, but you want to use asterisk IVR (cost, familiarity whatever) and want to add the feature of "direct dial extension" from that IVR.  The problem is FreePBX gui is designed to only look at the "Extension" list and use that to match.  Which makes perfect sense.  But if your asterisk box is only hosting an IVR, you wouldn't have all the extensions on the asterisk server.  This is basically a work around for that.

I've used this for DEMO purposes.  I wouldn't roll this code into production since its not been tested.  There could be all kinds of vulnerabilities!  Use at your own risk.

What the code does:
We modify the FreePBX IVR code not look at extensions, but to instead match the outbound dial plan of the Asterisk box that triggers calls to use the trunk to your downstream PBX

So if your PBX has 4000-4999 extension, you probably have a dial patter in an outbound route that looks like 4XXX

With this code, if someone in the IVR dial 4001, that will match this outbound route and send the call down the trunk.

1. Create an IVR with the “Direct Dial” option enabled in the GUI.  Note the name of it.  In my example its called IVR_Test

2. Save/Apply changes in FreePBX

3. Open up the file /etc/asterisk/ extensions_additional.conf

4. You want to look for the outbound route that you use to push calls to your downstream PBX Trunk under the [outbound-allroutes] heading.

(it’ll look something like this)
[outbound-allroutes]
include => outbound-allroutes-custom
include => outrt-16 ; PBX_PROD_01

Explanation of my “outrt-16 PBX_PROD_01” trunk
In my example above, outrt-16 is the outbound route in my gui that I use to trigger calls to send down a trunk.
If the number 4XXX is dialed, outbound route #16 “PBX_PROD_01” will be looked at to send calls down the trunk Called Asterisk_100_11  


5. So, copy the outbound route entry that matches your outbound route that’s used to trigger your Cisco trunk.  (my example you would copy   “include => outrt-16 ; PBX_PROD_01 “

6. Find the IVR in the code…it’ll look something like this:

[ivr-3] ; IVR_Test
include => ivr-3-custom
include => from-did-direct-ivr
exten => s,1,Set(TIMEOUT_LOOPCOUNT=0)
exten => s,n,Set(INVALID_LOOPCOUNT=0)
[[…..snip……]]
exten => h,1,Hangup
exten => hang,1,Playback(vm-goodbye)
exten => hang,n,Hangup
;--== end of [ivr-3] ==--;

7. Now paste the line you copied into the IVR and make it look like this:
[ivr-3] ; IVR_Test
include => ivr-3-custom
include => outrt-16 ; PBX_PROD_01
include => from-did-direct-ivr

8. Copy this IVR section and paste it into (Saving the IVR module in this override file ensures that future FreePBX changes won’t overwrite anything)
/etc/asterisk/ extensions_override_freepbx.conf
(Copy everything from [ivr-3] ; IVR_Test   right to ;--== end of [ivr-3] ==--; )
               No need to make any changes.

9. Save changes and close file
             
Don’t bother saving the changes in “/etc/asterisk/ extensions_additional.conf” as these get overwritten anyway by FreePBX.
You can just close this file without saving changes.
10. Now go into the console of asterisk using  “  asterisk –cvvvvvvr “
11. Type “RELOAD”
12. Now make a test call to your IVR and dial an extension.  See if it works!

Add HD and auto mount drive in linux (virtual)

Install the new VHD in your VM system.
Boot up the image.

fdisk –l

The circled item shows the new 100g drive that was added. "/dev/sdb"

FORMAT disk:

fdisk /dev/sdb

At the command prompt select " p " and press ENTER

Just double check to make sure the drive you are about to format name and size matches the one you installed

Now select option " N " and press enter

 

Now select option " P " for extended

The select Parition #1

For "FIRST SECTOR" and "END SECTOR" just press enter to accept defaults

Once thats done, select OPTION T

Select Partition 1 and then use option "L" to list your option codes to see them.

Select the option 83 (Linux) press ENTER

Now selection "W" to write your changes to the table

Now type the following:

mkfs –t ext2 /dev/sdb

That my take a couple of minutes for that command to complete.  When its done type in:

Type fsck /dev/sdb –f –y

The system should do some system checks, all of which should pass:

Reboot the system.

shutdown -r now

Once rebooted, we'll make a mounting point for this drive.  

mkdir /mnt/recordings

mount –t ext2 /dev/sdb /mnt/recordings

Now if you type "DF" you should see it in the list

type in the following:

cat /proc/mounts

look at the bottom, should see sdb reference

/dev/sdb /mnt/recordings ext2 rw 0 0

Might look something like this.  COPY that line then type

vi /etc/fstab/

paste the line above into the fstab and save it.  If yourdata has 'relatime' option.  change that to 'noatime'

save your changes..reboot

How to configure Static routing in Linux CENTOS

Static routes will be added usually through “route add” or “ip route” command. 

example:

route add -net DESTINATIONip netmask NETMASKip gw GATEWAYip dev ethX

So to add a specific route of 113.121.192.219 out gateway 10.128.216.129 via interface ETH1:

route add -net 113.121.192.219 netmask 255.255.255.255 gw 10.128.216.129 dev eth1

To make it persistent, add it to /etc/sysconfig/network-scripts/route-ethX

Adding Persistent static route:

MAKE SURE YOU HAVE THE SYNTAX EXACTLY RIGHT, ELSE IT WILL FAIL
CAPITALS!

You can have multiple ones of course, you just increment the value.  There is no space, follow the example below.  This one shows adding 3 different static routes.

GATEWAY0=10.1.100.1
NETMASK0=255.255.0.0
ADDRESS0=10.1.0.0

GATEWAY1=10.1.100.1
NETMASK1=255.255.0.0
ADDRESS1=10.2.0.0

GATEWAY2=10.1.100.1
NETMASK2=255.255.0.0
ADDRESS2=10.3.0.0

GATEWAY3=10.1.100.1
NETMASK3=255.255.0.0
ADDRESS3=10.4.0.0

Save and close the file and restart networking.  Reboot to prove!

Asterisk. Ask questions, record answers.

What this code does.

The system will ask questions, after each it will record the caller for a maximum of 15 seconds (you can change this of course) or until the users presses # key, whichever comes first.
This code allows for two questions and answers to be given.

In this example, after the 2nd answer is given, the system will play back both recordings together so you can hear it.

If caller press # the call will stop recording and continue to next questions

If caller either doesn't say anything for 3 seconds, the system continues to next question (changeable of course)

Once the questions are done, the caller is hungup on.

The system generates a .call file which will contact an extension of your choice, one that goes directly to voicemail.

The file that is recorded is saved to thevoicemail box, then the call hangsup.
Your asterisk system will send the email to the recipient using your voicemail parameters.

I've tested this in the FreePBX Distro using 13.0.190.11 worked ok as a beta test. It may need alteration for your own environments, however I am using commands that have been around a long time so should be compatible with most versions.

Code from strangers should never be tested in a production environment.

INSTALLATION
To make the code simpler, for those not versed in asterisk dialplan, any line starting with a ; semicolon is a remark line and can be safely removed for ease of viewing. Lines with ; are there to either leave a comment, or to have the system not execute that line of code.

STEP 1
ADMIN->CONFIG EDIT
open up "extensions_custom.conf"
Copy the code below between the ------ lines into the bottom of extensions_custom.conf and then save the file.
Replace code from previous BETA 1 & 2 releases

   START OF DIALPLAN CODE
   ---------------------------------------------------------------------------

[macro-questionanswer]
exten => s,1,answer()
exten => s,n,playback(/var/lib/asterisk/sounds/en/custom/question1)
exten => s,n,Record(${UNIQUEID}.ulaw,3,15,a)

exten => s,n,playback(/var/lib/asterisk/sounds/en/custom/question2)
exten => s,n,Record(${UNIQUEID}.ulaw,3,15,a)

;Generate .call file that asterisk will use to send the call to a voicemail box

;Change XXXXX to be an extension in your system that you want called and have the message left on

exten => s,n,system(echo "Channel: LOCAL/XXXXX" > /var/lib/asterisk/sounds/recording.call)

exten => s,n,system(echo "Context: macro-questionanswer1" >> /var/lib/asterisk/sounds/recording.call)

exten => s,n,system(echo "Extension: s" >> /var/lib/asterisk/sounds/recording.call)

exten => s,n,system(echo "Setvar: playfile=${UNIQUEID}" >> /var/lib/asterisk/sounds/recording.call)

exten => s,n,system(echo CallerID: "${CALLERID(name)}" ${CALLERID(number)} >> /var/lib/asterisk/sounds/recording.call)

exten => s,n,system(mv /var/lib/asterisk/sounds/recording.call /var/spool/asterisk/outgoing)
exten => s,n,hangup()

[macro-questionanswer1]
exten => s,1,wait(1)
exten => s,n,senddtmf(#)
exten => s,n,playback(/var/lib/asterisk/sounds/${playfile})

;delete sound file.
exten => s,n,system(rm /var/lib/asterisk/sounds/${playfile}.ulaw)
exten => s,n,hangup()

  -----------------------------------------------------------------------------
  END OF DIALPLAN CODE

Click on "SAVE"

If you are updating this code from previous versions, you can stop here and test it out.
SCROLL DOWN TO THE BOTTOM TO THE "TESTING IT OUT" section

STEP 2
ADMIN->CUSTOM DESTINATIONS
Create a new custom destination "questionanswer"
Set the target as
macro-questionanswer,s,1
Save those changes

STEP 3
Create a way to trigger the application to run from dialplan.
For example, create a IVR, and set one of the numeric values a caller can enter to Custom Destination -> "questionanswer""

STEP 4
Save your changes AND apply the configuration.

STEP 5
Create your question files.

Use system recordings, and either record them over your phone, or through a 3rd party sound recording application. If recording with another app, they must be saved as 8000khz MONO 16bit WAV PCM.

The recordings you make, set their name to "recording1" "recording2" "recording3"

STEP 6
in /tmp/ create a file called "email.txt"
In that file put in something like
SUBJECT: NEW RECORDING
Here is a new recording to review.

Save changes.

TESTING IT OUT

Call and leave your recordings. System should leave the message and you should get an email alert with the message attached.

HOW IT WORKS
1> As previous, your recorded questions are played, and the answers are recorded into a single file.
2> After the last question is answered, the system will hangup.
3> Then the system internally places a call to the extension that you want the system to send the voicemail through.
4> When that system picks up, assuming voicemail, the system sends a "#" tone to the extension to bypass any greetings and additional instructions that voicemail systems have.
5> The system in the background plays the message that your caller left, and sends it as a voicemail.
6> Your recipient should get the message alert in their email.

Give it a test!

.CALL FILE EXPLAINED.

If you create .txt file with the proper calling parameter information, you can drop that file in the outgoing folder in asterisk and that will make your phone call happen.
Essentially these 4 lines generate the 4 line file.

The number you want to call, this example is extension 1234
exten => s,n,system(echo "Channel: LOCAL/1234" > /var/lib/asterisk/sounds/recording.call)

The context you want executed when the destination 1234 answers
exten => s,n,system(echo "Context: macro-questionanswer1" >> /var/lib/asterisk/sounds/recording.call)

The line you want executed in the context
exten => s,n,system(echo "Extension: s" >> /var/lib/asterisk/sounds/recording.call)

Variables you want passed to the system when 1234 answers
exten => s,n,system(echo "Setvar: playfile=${UNIQUEID}" >> /var/lib/asterisk/sounds/recording.call)

The variable is the most complicated piece. UniqueID is the unique number given for each call. When you are recording the calls, we use that number as the file number. However, when we call the new number to leave the voicemail, that generates a new unique ID. So in this code we set a temporary variable of "playfile" to store the unique number that is the file name. This temporary file is, in itself unique to this call, so it wont affect things if you have multiple question answers recording simultaneously..

Asterisk Trouble Shooting

Recently saw this appearing in my ASTERISK Live LOG over and over

netsock2.c: getnameinfo(): ai_family not supported

There are a few entries on the internet pointing to IAX trunks.  Mine was related to IAX.  I had created an IAX trunk on the server in questions, but not finished it properly.  In my case I only created one end of it, the other PBX was not done, so just started popping this message over and over.  Deleted the trunk, all was resolved.