Michael McNamara

technology, networking, virtualization and IP telephony

Avaya 1100 Series IP Phone Upgrade to SIP

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Over the past weekend I set out to setup Asterisk, an open source communication server, to test some of the issues reported in a thread over on the discussion forums. I had an Avaya 1120e and 1165e IP phone available to test with, however, both phones were running the UNIStim software for the Avaya Communications Server 1000. I needed to upgrade them to support SIP.

About a year ago I posted how I setup Asterisk to work with an i2002 IP phone utilizing the UNIStim channel driver. This time around I was looking to utilize the standard SIP channel driver with the 1120e and 1165e.

SIP Software

You’ll need to download the SIP software from the Avaya Support website. You should be able to retreive the SIP software from this link without needing to log into Avaya’s website. You should download the software for the appropriate model you’ll be working with. In my case I downloaded the following two files;

  • SIP1120e04.00.04.00.bin
  • SIP1165e04.00.04.00.bin

TFTP Server

You’ll need a TFTP server to host the files that the IP phone will download. You can use any TFTP server you already have on the network. If you don’t have a TFTP server you can use TFTPD32 from Philippe Jounin on any Microsoft Windows XP, Vista or Windows 7 personal computer. I download the zip and exploded the files to D:\Temp.

TFTP Files

With the TFTPD32 software in D:\Temp I then copied the two firmware images (SIP1120e04.00.04.00.bin and SIP1165e04.00.04.00.bin) to the same directory. At this point I needed to create some configuration (provisioning) files which the IP phones would download. The first file 1120e.cfg will be used for the 1120e IP phone;

[FW]
DOWNLOAD_MODE AUTO
VERSION SIP1120e04.00.04.00
FILENAME SIP1120e04.00.04.00.bin
PROTOCOL TFTP
SERVER_IP 192.168.1.3
SECURITY_MODE 0

I also created a file 1165e.cfg that would be used for the 1165e IP phone;

[FW]
DOWNLOAD_MODE AUTO
VERSION SIP1165e04.00.04.00
FILENAME SIP1165e04.00.04.00.bin
PROTOCOL TFTP
SERVER_IP 192.168.1.6
SECURITY_MODE 0

You’ll need to substitute the IP address above (192.168.1.6) with the IP address of the personal computer that will be running TFTPD32. Now that you have all the files you’ll need for the upgrade, you can start the TFTPD32 executable. You should see a window similar to the figure to the right.

Upgrade

You need to make sure that the IP phones know which TFTP server to use. This can be accomplished via DHCP option 66 or it can be set in the device configuration on the actual IP phone itself. I was utilizing the DHCP server built into my Verizon FiOS router so I had to set the TFTP server manually via the IP phone configuration.

When you are ready just reboot the phone. As the IP phone boots up it will request an IP address from the DHCP server and it will check the TFTP serve. The IP phone should download the 1120e.cfg (or 1140e.cfg of 1165e.cfg depending on the model). Once the phone realizes there is a software update it will boot into BOOTPC mode in order to perform the actual upgrade.

You should see something similar to the following;

[FW] reading...
SIP1120e04.00.04.00.bin
VERSION SIP1120e04.00.04.00

Shortly followed by;

[FW] writing...
SIP1120e04.00.04.00.bin
VERSION SIP1120e04.00.04.00

Once the upgrade is complete the IP phone should reboot. I will warn you that you should I’ve seen some odd behavior between the settings on the IP phone and the settings that should be applied via the provisioning files. There have been a few cases where I needed to reconfigure the IP phone even though it appeared to be configured properly. In the few cases I’ve experienced reconfiguring the IP phone solved the problem.

Once the 1100 series IP phone is upgraded to SIP it will start looking for a new configuration file, 1120eSIP.cfg (or 1140eSIP.cfg or 1165eSIP.cfg depending on your model).

Here’s a quick example;

[FW]
DOWNLOAD_MODE AUTO
VERSION SIP1120e04.00.04.00
FILENAME SIP1120e04.00.04.00.bin
PROTOCOL TFTP
#SERVER_IP 192.168.1.3
#SECURITY_MODE 0

[DEVICE_CONFIG]
DOWNLOAD_MODE FORCED
VERSION 000002
FILENAME users.dat

[DIALING_PLAN]
DOWNLOAD_MODE FORCED
VERSION 000002
FILENAME dialplan.txt

Here’s a copy of the users.dat file which gets called from the 1120eSIP.cfg file above;

DNS_DOMAIN asterisk.home
SIP_DOMAIN1 asterisk.home
SERVER_IP1_1 192.168.1.6
SERVER_PORT1_1 5060
SERVER_RETRIES1 3
DEF_USER2 ASTERISK
VMAIL 5000
VMAIL_DELAY 300
DEF_LANG English
DEF_AUDIO_QUALITY High
ENABLE_LLDP YES
ADMIN_PASSWORD 26567*738
ADMIN_PASSWORD_EXPIRY 0
# Settings to disable extended license
MAX_LOGINS 1
USB_HEADSET LOCK
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
# Enable SSH
SSH YES
SSHID admin
SSHPWD admin

The settings above disable any advanced features and allow the IP phone to run a basic SIP configuration.

Cheers!

You might also want to read these other posts...

  1. Asterisk Now with Avaya IP Phones
  2. Avaya 1200 Series IP Phone Configuration Options
  3. SIP Software Release 4.3 for 1100/1200 Series IP Phones

Comments

  1. Hi Michael,
    how to change date and time on nortel sip phone? I had it connected to asterisk.

  2. I have phone setup per the instruction at beginning of the post. phone have token request: no request, but the day still count down. Any one have this issue, or it’s just normal. FYI.

    • Hi Steven,

      As long as it says “no request” you should be good. It might be that once you enable any of the features the demo license will start it’s countdown regardless if the features are no longer being used.

      Cheers!

  3. Is there any known Avaya reseller to buy small qty of extended SIP licenses from? What do they cost approximately? I bought an expansion module and I also want to use Bluetooth… so I need some.

    Thanks!

  4. I don’t believe legacy Nortel products use comcodes, those are the pure Avaya components such as the Definity and Partner systems. Am I correct Michael? The sip Licenses are extremely hard to obtain, as I have tried and most Avaya vendors have no idea what I am talking about. People who deal with communciation server 1000 and 2100 are your best bet in my book.

  5. Hi Michael,

    Have you ever setup Intercom feature on this series phone? I’m stuck at how to program a soft key to be Intercom key so that when I press that key and enter a DN, it originates an Intercom call (with “Answer-Mode: Auto Answer” appears in INVITE msg) to that DN.

    BTW, I’m using SIP software 04.01.15.00 for my 1140.

    Thanks in advance,
    Tien

    • I was able to do this after days of searching using:

      exten => 1000,1,SIPAddHeader(Answer-Mode: auto)
      exten => 1000,n,Dial(SIP/nortel)

      Also in the phone features – auto answer you have to whitelist the source. I put in the ip for my asterisk box and it worked fine. I think I left the first option at disabled but put in the ip on the 2nd option.

  6. Hi Michael,

    I’m using this phone with SIP application server A2 of Genband. The A2 version 9.0 supports Intercom feature but I don’t know how to program the Intercom key on the Avaya phone.
    BTW, I’m a system verification engineer of A2.

    Thanks,
    Tien

  7. Hi Michael,

    Excellent guide…i followed through to upgrade my 1140E successfully. Is it possible to configure this 1140E to work with BroadVoice? I was able to configure Cisco 7940 to work with BroadVoice…(BroadVoice has a tftp server for Cisco 7940 to connect with login info configured in the configuration section).

    Thanks,

    Jay

    • Hi Jay,

      I can’t say… you’d probably have to test with BroadVoice or inquire with them. There have been reports from a number of followers that they’ve been able to connect the 1100 and 1200 series IP phones with third party SIP providers. I believe there was a thread or two in the Avaya IP Telephony forum but I don’t have the time right now to try and track them down.

      Cheers!

  8. Dear Sir,
    we have Nortel 1120E phone and want to upgrade to SIP…….we could see a error code 10054 (an existing connection was forcibly closed by the host)………….

    Need your help

    • Hi Manjunath,

      You’ll need to give me some more detail… you get that error where? At what step in the process? Do you see the connection to the TFTP server? Do you see the IP phone downloading the 1120e.cfg file?

      Good Luck!

  9. Hi Michael,

    I told you about that new Asterisk system called UCx from Emetrotel. Well I am now testing one out at home and the Unistim versions of Nortel phones are working great.

    I’ve also connected 4 SIP sets, Nortel 1140E, Nortel 1120E, Cisco7970, and Avaya 9630 to my Call server and they work just as well as the Unistim sets, except that MWI is a continued problem.

    On my Nortel 1140E and 1120E, (I am upgraded to the latest 4.03 firmware with No tokens requested) the MWI light appears 30 minutes after a Message is left. After deleting a message, it appears that the message light takes another 30 minutes to dissappear. Restarting the phone takes care of the problem right away and the Light either appears or shuts off depending on the situation.

    Any thoughts?

    On the Cisco 7970 and Avaya 9630 SIP, nothing happens with MWI.

    Joe

    • Hi Joe,

      I would probably suggest that you run a packet trace against the 1120e/1140e and try to determine who’s at fault, the IP phone or the Call Server.

      You’ll be looking for the SIP NOTIFY or SUBSCRIBE message that include the MWI flag. If you don’t see any packet MWI messages until 30 minutes later you’ll know the issue is with the Call Server (or more likely the actual Voice Mail application).

      http://www.ietf.org/proceedings/48/SLIDES/sip-waiting.pdf

      Good Luck!

  10. Hi Michael,

    Than you for your response!

    It looks as though the Call Server is at fault.

    I have a Cisco 7970 and Avaya 9630 both connected as SIP phones as well, and they do not respond at all to the MWI, which really also makes me believe it’s a Call Server issue.

    I am having my friends at Emetrotel look it over….

    I’ll let you know what happens.

    I have Unistim phones MWI instantly responds to the MWI light, but not the SIP :(

    Cheers,

    Joe

  11. Michael,

    Alright……I made some progress this evening :)

    I referred to this link:

    http://www.freepbx.org/forum/freepbx/users/mwi-in-1-6-2-not-working-possible-sip-configuration-issue

    I recreated the SIP extensions on my Asterisk switch and had 3 results with my 3 different phone types:

    Cisco 7970 SIP phone works flawlessly-the light comes on when a message is received and shuts off when I exit Voice Mail.

    Nortel 1140E and 1120E MWI light turns on when Voice Mail is left….BUT DOESN’T TURN OFF WHEN MESSAGE IS DELETED AND VOICE MAIL IS LOGGED OUT….I am pulling my hair out over this one!

    I made 4 successful test messages with the MWI light popping on when I received a message on the 1140E, but then in order to remove the MWI, I have to reboot the phone. So something is keeping it from releasing the MWI and I have no clue.

    Avaya 9630 phone is completely unresponsive, even when I deleted the extension and rebuilt it. I don’t have much experience in this phone set. I assume this set doesn’t function to well on a 3rd party server.

    Let me know if you know a way for me to get the MWI light to turn off on the 11xx sets!

    Thank you,

    Joe

  12. Even more progress as I go to sleep now. The 1140e I have connected to SIP works perfect now, as I recreated it’s a new extension order the link I sent. The red light comes on when I leave a message and turns off when I press the goodbye or rls key. If i hang up using # the light doesn’t shut off.

    I also have an 1120e and did the same procedure of deleting it and restaring it and the MWI light comes on for a new message but unlike the 114Oe the red light doesn’t shut off…… Very frustrating!

    Any thoughts?

    How

  13. We have nortal 1120 Series of IP phones…..im planning to change my IP Server to Siemens Openscape MX which is SIP Compatable. Please help me to make my nortal 1120 IP Phones to upgrade to SIP enabled for me to use the same phone with Siemens MX. Since nortal support is not available in india, is it possible to upgrade all my phones to SIP version without any cost???

  14. Michael,

    I am stumped. I have 2 11xx phones running Sip connected to an asterisk UCx from Emetrotel. As mentioned before, the MWI light keeps acting weird. On one extension I had the MWI light working perfectly while on another extension it didn’t work right, as in once you receive a message and listen to it, MWI never shuts off. Sometimes it doesn’t even come on at all?

    Is there something in the code I can change? The call server seems to be working fine….I have a Cisco 7970 connected and it works perfectly all the time.

    Thanks again,

    Joe

  15. I am trying to configure the 1120e and 1140e for auto login.

    I have files
    1120e.cfg
    1120eSIP.cfg
    SIP001ECAFEC786.cfg
    001ECAFEC786.cfg
    user.dat

    on the TFTP server.

    The problem, the phone only reads the 1120eSIP.cfg file and nothing from the SIP001ECAFEC786.cfg file. Can someone enlighten me as to the contents of this file.

    I made it the same as the 1120eSIP.cfg file with the exception that it should load the 001ECAFEC786.cfg file instead of the user.dat file.

    Stuck here for ages now.

    • Hi Obasi,

      When working with provisioning IP phones the old UNIStim firmware would look for system.prv, .prv and then .prv in that order.

      I’m not sure of the behavior of the SIP firmware, although I’m not sure that you should care yourself (unless you’re trying to automate a few hundred IP phones).

      You can log into the IP phone manually the first time and it will store the username and password (by default) for subsequent reboots. So on subsequent reboots the IP phone will just connect to your SIP provider.

      Hopefully that helps.

      Cheers!

      • This is how we have them now, but I would like to write something small to manage them remotely.

        But thanks for the assistance. I will play with the unit on the desk some more.

        Cheers :)

      • I have gotten the Auto provisioning to work :)

        1120eSIP.cfg (Add)
        [USER_CONFIG]
        DOWNLOAD_MODE FORCED
        VERSION 0101
        PROTOCOL TFTP

        Create a file: SIP.cfg ( MUST BE IN CAPS)
        Add everything from users.dat, and also
        AUTOLOGIN_ENABLE USE_AUTOLOGIN_ID
        PROMPT_AUTHNAME_ENABLE NO
        AUTOLOGIN_ID_KEY01 @
        AUTOLOGIN_AUTHID_KEY01
        AUTOLOGIN_PASSWD_KEY01

        is from this line
        DNS_DOMAIN

        Reboot your phone and smile :)

  16. Some of my text seems to have gone missing:

    1120eSIP.cfg (Add)
    [USER_CONFIG]
    DOWNLOAD_MODE FORCED
    VERSION 0101
    PROTOCOL TFTP

    Create a file: SIP.cfg ( MUST BE IN CAPS)
    Add everything from users.dat, and also
    AUTOLOGIN_ENABLE USE_AUTOLOGIN_ID
    PROMPT_AUTHNAME_ENABLE NO
    AUTOLOGIN_ID_KEY01 [username]@[domain]
    AUTOLOGIN_AUTHID_KEY01 [username]
    AUTOLOGIN_PASSWD_KEY01 [password]

    [domain] is from this line
    DNS_DOMAIN [domain]

  19. Hi Michael,

    Update on the MWI light, it now works on the SIP 11xx and 12xx phones on the Emetrotel UCx Asterisk platform! The problem was a bug in the Asterisk software…..

    JOe

  20. HI michael:

    I am have an Avaya 1210 Ip deskphone, the phone show in the display

    “Repair of File System” in the firth line, and the second line show ” Wait a few minutes”

    thank you, sorry for my languaje. i am speak spanish.

  21. A mi me pasa lo mismo con un 1210, en pantalla tengo el mismo mensaje: “Repair of file system”.

    ¿Has resuelto la incidencia?.

    Saludos.

  22. Hello,

    I’ve downloaded the SIP 4.0 file from this thread and when attempting to upload the firmware, the phone states that it is reading the firmware and returns with [FW] Auth. fail.

    I have checked and my 1140e set is in the approved list for the 4.0 SIP firmware, but cannot figure out what this means. I changed the download mode to forced to see if that would make a difference but it does not. The TFTP correctly downloads the file without failure.

    Thanks,
    Paul

    • Hi Paul,

      What version of firmware is currently loaded on the IP phone?

      I would probably advise re-downloading the SIP firmware again (perhaps it was corrupt?)

      Good Luck!

      • the current version is 0625C8A on UNISTIM. i’ll try downloading from the link above again and see if that helps.

  23. Your instructions for upgrading my Nortel made 1120e phones was great. My problem now is that I have no idea how to configure the phone with actual values. When using X-Lite is simple and straightforward. Going through the endless values on the 1120e is driving me crazy. I am using the instructions from Avaya for the SIP version I have and still I have NO clue if my values are correct.

    What else do I need?

    my users.dat file looks like this:

    DNS_DOMAIN telops.ccgvca.com
    SIP_DOMAIN1 telops.ccgvca.com
    SERVER_IP1_1 209.58.229.12
    SERVER_PORT1_1 5060
    SERVER_RETRIES1 3
    DEF_USER1 119
    ENABLE_SERVICE_PACKAGE YES
    ENABLE_3WAY_CALL YES
    TRANSFER_TYPE STANDARD
    REDIRECT_TYPE RFC3261
    FORCE_BANNER YES
    BANNER Classroom 1
    UPDATE_USERS NO
    ENABLE_UPDATE YES
    ENABLE_PRACK YES
    RTP_MIN_PORT 50000
    RTP_MAX_PORT 50100
    SIP_PING YES
    AUTOLOGIN_ENABLE YES
    DEF_LANG English
    VMAIL *98
    VMAIL_DELAY 300
    EXP_MODULE_ENABLE YES
    ENABLE_BT YES
    DST_ENABLED YES
    TIMEZONE_OFFSET -28800
    SNTP_ENABLE YES
    SNTP_SERVER pool.ntp.org
    AUTO_UPDATE YES
    AUTO_UPDATE_TIME 3600
    AUTO_UPDATE_RANGE 1
    MAX_INBOX_ENTRIES 50
    MAX_OUTBOX_ENTRIES 50
    MAX_REJECTREASONS 5
    MAX_CALLSUBJECT 5
    IM_NOTIFY NO
    IM_MODE DISABLED
    DEF_DISPLAY_IM NO
    MAX_IM_ENTRIES 20
    MAX_ADDR_BOOK_ENTRIES 100
    ADDR_BOOK_MODE BOTH
    RECOVERY_LEVEL 0
    ADMIN_PASSWORD 26567*738
    DISABLE_PRIVACY_UI YES
    LOGOUT_WITHOUT_PASSWORD NO
    NAT_SIGNALING NONE
    NAT_MEDIA NONE
    HOLD_TYPE RFC2543

    # Enable SSH
    SSH YES
    SSHID admin
    SSHPWD admin

  24. For the most part, this worked very well!

    I used the SIP firmware version 2.2 with an 1120e because I didn’t want to deal with the advanced/basic licensing issue, it seems that almost everything is working perfectly!

    The only problem I have is that the phone does not alert me to when new voicemail’s arrive in any way. No 1 new voicemail, nor a light of any kind.

    I pretty much used your config files, and kept the default voicemail of 5000 in the users.dat file. It doesn’t seem to work that way, so I manually changed it to *97 in the phone’s options; but still no dice. It works perfectly on my Cisco 7960 but of course, that’s a completely different phone.

    Running PIAF with Asterisk 1.8

    any suggestions?

    Thanks!

    • Hi Blake,

      The entry in the users.dat file is just the DN to be used to dial voicemail should someone push the Inbox key. You are referring to a Message Waiting Indicator (MWI) which I’ve never personally tested.

      You might want to post over the discussion forums where your question might get some better exposure and perhaps an answer from someone other than myself.

      Good Luck!

    • From my Engineer at E-Metrotel, we fixed the MWI problem on my SIP phones running on the UCx50 Asterisk based Call Server.

      “Specifically for the fix I did for the MWI, it has been just made available in the Asterisk version 1.8.17.0-rc1 on Sep 13. Typically, the release candidate is converted into a standard release in a month or so. You can advise the person that a fix for this problem will be available in Asterisk 1.8.17.0.”

      So update to 1.8.17.0 and your MWI problem should be fixed. My 1140E and 1120E sip MWI is working perfectly.

      Blake, have you thought about using the Unistim Channel Driver and running the phones on Unistim? They operate a lot more easier on the Unistim platform than they do on SIP, and they also come with a ton more features on Unistim.

      Just curious.

      Joe

  25. Managed to get the sip bin file onto my 1140e and it does seem to log into the sip account as I can make outgoing calls, but I’m not able to receive calls (on the sip number, the extension or direct ip call). Any idea why?

    Thanks.

    • Hi Wenkie,

      You’ll need to provide some additional information… where’s your Call Server? Are you running your own instance of Asterisk? Is it on the same network as your 1140e IP phone? Have you check the logs of your Call Server to see if the call is even coming to your system? Are you placing the call over PSTN or SIP trunks? So many questions so little information to really be of any help.

      Good Luck!

      • Hi Michael

        Thanks for your response. Not sure about the call server bit – we’re using hosted voup (soho66.co.uk). Setting up the old grandstream was very simple (enter server, user and pw), similar experience with softphones on the pc. New to these avaya phone, but was expecting something similar.

        Thanks.

  26. Hi Michael,

    I know it’s been awhile when you posted this blog entry but, since I’m unable to find answers on the web, I decided to take a chance here and ask my question to you directly.

    I’m working at migrating quite a big user base of 1120e/1140e from UNIStim to SIP and everything is working smoothly so far. Thanks for all your tips by the way. So I got all my provisioning files and processes in place and it is now possible to do the migration. There’s only one small thing… remote reboot.

    Here’s the question: is it possible to remotely reboot those UNIStim phones? I want to avoid any manual intervention and I know that once the phones have reboot, the whole process kicks in and the migration becomes flawless.

    I tried to look at what’s available on the MCS 5200 that is in place and the only option available that looks like a reboot is the reset function. Unfortunately, it does not perform a reboot, only a reset of the connection to the network. What I really need is the phones to reboot so they will get their new provisioning server and will download the new config files to upgrade.

    I also looked at what’s available using SSH, thinking that I could script a little something. With the SIP firmware, I know the reset2factory function is available and that’s what I was thinking of using. Unfortunately, the UNIStim firmware that I have on those phones is now quite old and the function doesn’t exist. I know there is a sendKey function that I thought could be useful but I can’t find any information on what can be sent to the phones using this method. Did you try to use this function?

    If you can think of anything else, that’d be really appreciated!

    Thanks for your help!

    • Hi Jean,

      I can think of a number of way to cause the IP phones to reboot…

      1) You could push a new firwmare (umsUpgradeAll) via your Signaling Servers and Call Server. That would cause the IP phone to reboot to perform the upgrade, however, I believe it will follow the DHCP and TFTP provisioning files instead of the UTFTPD update process.
      2) You can instruct the IP phone to reboot from the Signaling Server, you could build an Except script to login to the Signaling Server and dump a list of IP phones and then issue the commands to reboot each IP phone one at a time.
      3) Assuming these phones are connected via PoE switches you could cycle the power from the PoE switch.
      4) Disrupt the connectivity to both Signaling Servers and the Call Server. If you disrupt the networking to/from the IP phones and the Signaling Servers the IP phones will watchdog timeout and reboot in an attempt to recover. You could accomplish this by using some ACLs to intentionally block traffic to/from specific subnets or you could literally shutdown the ports connecting your Signaling Servers.

      Good Luck!

      • Thanks Michael! These are great tips that I’ll investigate.

        Since I’m still curious, do you have any experience with the sendKey function available in PDT? I tried to map the various values and I realized that those have changed from release to release (at least, they are not the same when going from UNIStim to SIP).

        Thanks

        • I’m not familiar with the sendKey function. You should have everything you need from the Signaling Server CLI interface if that was to be your approach. Probably much safer too, not sure I would advise running a script within PDT – dangerous.

          Good Luck!

  27. hey,

    Im also getting the [FW] Auth. Fail issue.

    Tried redownloading the file but still the same.

    Any other ideas?

  28. Getting nowhere with this. I”ve got an 1140e running unistim 0625C8J and I just can’t get the phone to see my TFPT server. It keeps “starting DHCP” Where in the settings do I put the TFTP server address. I jus don’t get it.

    • You can set the TFTP server on the phone in settings – it’s towards the bottom of the options and I think it’s name changes based on the firmware revision but look for Provisioning Server. You’ll need to set DHCP to Partial (or off)

    • Also check your settings on the phone and make sure there aren’t a bunch of “ffffff” under S1 and S2, if there are try changing to a string of “121212…” (16 I believe but just keep entering until it stops). I’m not positive on this but it’s helped me, also turning off Voice VLAN and Data VLAN helped (there’s checkboxes saying something like Disable Voice Q## and one for data as well, some versions say Enable, some Disable – either way try having them disabled)

      Hope this helps and makes sense, I’ve only worked on these phones a few times and this is my Sunday morning memory you’re getting :-)

  29. Hi all
    i’ve a big problem!

    i’ve 1210 Tel and i’ve upgraded to SIP using this SIP12x004.03.12.00 and i’ve used Asterisk as PBX, every time that someone call my Avaya 1210 phone the phone restart without say more, i’ve tried this on more than one tel, but i’ve the same problem.
    any suggestione?
    thanks
    Giovanni

    • It’s happening because phone sends SIP PING packets, but Asterisk doesn’t answer.
      Try to add line SIP_PING 0 to users.dat

    • Hi Giovanni,

      Not sure if you’ve resolved the issue yet, but I’ve found that using any firmware newer than 03.02.16 on the 1210 phones introduces the problem you are having – roll back to that version and you won’t have that issue anymore. Not exactly a solution, I know, but that’s the way to go if you need the phones up and running ASAP.

  30. I currently have four 1140E phones that I was told should work with Avaya IP Office. The load version is 04.03.12.00 with software of SIP1140. I have done nothing to the software of these phones (I usually work with Avaya IP phones so this is a completely new thing for me). I plugged the phones in after creating the user + extension and I hit the login screen. I log in with the created user information and I go right into the home screen for the phone. I look at the extension in the IP Office and I see it is being detected as an Avaya 1140E SIP phone so all good there.

    My problem is after I log into the phone (any of the four phones, not restricted to one in particular) after roughly 30 seconds to 1 minute, the phone logs out and back in on its own, disconnecting the user from anything they were doing on the phone.

    I’m not sure if you’ll be able to give any insight into my issue but I’m finding little information elsewhere so figured I would ask.

    • I only just recently got my hands on an Avaya IP Office. What version of software is the Avaya IP Office running?

      It’s sitting in the cubical across from me along with a few Avaya 9600 Series IP phones.

      I’m going to guess that there’s some timeout occurring. You might get a hint to the problem by performing a packet trace on the 1140E to see what’s being said between the handset and the call server.

      Sorry, I can’t offer more help.

      Cheers!

  31. Hi Mike,
    i purchased one of this particular Avaya 1140E. I prepared the cfg File according to your Instructions. 1140e.cfg. I use TFTP32 for upload. I configured a DHCP Server on TFTP32
    in the Logiles I can see the Phone has been assigned an IP Address. Then it is searching for CFG Data. then it goes over to system.prv and says failed. Finally it is looking for the TFTP Server which also failes. All Files needed are correctly placed into the Root Folder.
    Im also not able to enter the Menu to configure the TFTP IP Address manually
    Thanks a lot for Assistance.
    This is how it looks like.
    [FW]
    DOWNLOAD_MODE AUTO
    VERSION SIP1140e04.03.18.00
    FILENAME SIP1140e04.03.18.00.bin
    PROTOCOL TFTP
    SERVER_IP 192.168.1.100
    SECURITY_MODE 0

  32. Hi Mike,

    finally I was able to upload the Firmware on the Phone. Now also the Settings on my Cisco Router have been corrected. One More Question; how do i change the Phones Time Mode from 12hour (PM) to 24 hour Mode?
    This are my Settings.
    DST_ENABLED YES
    TIMEZONE_OFFSET 60
    FORCE_TIME_ZONE YES
    Already time looks ok, but 4:00PM
    Thanks for your Help.
    Best Regards
    Andy

  33. At first great thanks for all you posts about nortel phones setup. It’s very helpfull.

    If anyone try to update from old firmware (like 0624C1B) – do not try to use vlans. Update process hangs up on trying to get address from dhcp. So use untagged ports and all will be fine.

  34. How know how to configure BLF 1140/1120/1220/1230 SIP firmware?
    or where i can find and example the sip documentation is very small

  35. Hi Michael

    We just went from CS1000 to IP Office 10 and we decided to keep all our 1120 and 1140 phones and just upgrade the firmware to SIP. This all went pretty well except we lost a bunch of features when doing this. Do you know if we are able to do the following with the new SIP firmware:
    1. Corporate Directory
    2. Key Expansion Modules
    3. Access voicemail from the messages button.

    I guess what I am asking is if you know of all the features that can work with IP Office. :-)

    Kind Regards
    Bradley

    • Hi Bradley,

      Yes you’ve found SIPs dirty little secret, there are a lot less features in SIP than in the legacy H.323 phones.

      I’m sure there’s some documentation somewhere from Avaya that lists the SIP capable features in IP Office.

      Good Luck!

  36. Dear Michael!
    Finally I could convert my ip phone 1120e to SIP, but I have got some dudes:

    “Licensing Info” show me:

    License Tokens
    License mode: Node locked
    License status: No request
    Token expiry: 31 day(s)
    Token type: Evaluation
    Tokens requested: 0
    Tokens acquired: 0

    AND

    On “Licensable Features” there is not selected nothing.

    Is fine? or when 31 days is over, the ip phone will not work?

    and last.. How can I activate the Port PC of ip phone?, I tried anything, but I cannot do it, just works the principal port of ip phone, I mean, only the PoE Ethernet.

    Thanks you for your reply!

    Sorry for my english!

    From Chile! Alejandro!

  37. Nothing better then commenting on old posts. Michael, are you aware of any way on the 11xx sip phones to pull all the current button configurations/users. Im currently scripting something that will ssh to the phone and key through each button and pull the line or speed dial information. Since one is able to go to prefs > feature options > feature keys and see what each key has on it, perhaps there is a way to pull that info, maybe as a xml file.

    Also, do you have any idea on how to access the vxshell on the 11xx phones. when you try and go there it brings up a challange and site to visit to get a response, but that site appears dead.

  38. Hi Michael.

    I have 20 AVAYA 1210 phones from the Nortel Legacy line, but I need to install firmware for SIP protocol, I have searched AVAYA but I have not been lucky, I understand that it is discontinued, so I wanted to ask if in your files you have some old firmware SIP protocol for this model.

    I would appreciate you help.

    Best Regards!

  39. hi micheal
    thanks for your educated posts.
    i wish you can help me with the following issue, my phone (1140e) was working just fine with sip protocol and asterisk server, one day after a reboot, i got the following message “Updating file system wait a few minutes”!
    and the message still appear on my screen! how can i solve this issue?
    thanks in advance

  41. Hi Michael,

    Sorry for the dumb question, I have a AVAYA 1230 phone and I successfully changed firmware to latest SIP firmware. I’m following your steps to provision the sip phone but stuck on users.dat file. Is there a way to supply my voip.ms sip credential in the file so when the phone boots up, it registers on voip.ms directly so I can make phone calls immediately. I tried to login on the phone but it didn’t work.

    Thanks,
    David

    • Hi David,

      Were you successful in logging into voip.ms and your just looking to automate the process? Unfortunately, I haven’t done any testing with the Avaya 1200 series in quite a few years. Sorry.

      Cheers!

      • Hi Michael,

        Thanks for your reply:-)

        I can login to voip.ms using my sip credential manually on the phone, but I’m looking for a way to automate this process by adding some lines to users.dat file. I googled the instructions on how to change users.dat file but couldn’t find anywhere.

        Is there a way to change the text for each line key? Currently my phone displays truncated telephone numbers because it only allows maximum 9 characters. Also, I’m wondering if you can post some dial plan example.

        Thanks,
        David

  43. I realize this post is really old so don’t know if it’s still active. Here goes. I have a need to to convert a number of 1140e sets from UNISTIM to SIP. I have done one set and the conversion was successful with the help of some of your instructions. So here is my problem; I want these phones to register to my Audiocodes Mediant 3000 SBC as Far End Users. I already have quite a few, 20-30 Polycom IP-7000 sets and a couple of Audio Codes 440HD registered and they work properly. In your sample files I don’t see where you specify the sets Directory number or how to configure the set for TLS and SRTP. You also make reference to a “dialplan.txt file but no examples. I do have these in the Polycom but I doubt that they would be the same. When the set comes up now it askes for the ID and password I have put the set DN in as the ID (44737), but am not sure about the password. I have entered 44737 here but the set goes to another screen and finally just says “logging in user” and just stays there. Looking at the Sys Logs in the M3000 I see the set trying to register but always fails. So at this point I’m not sure iwhere the problem lies or even if the M3000 will support the 1140e. The LW version on the 1140e is 04.04.29.00.

    • Hi Steve,

      Unfortunately I haven’t worked with those IP phones in a very long time…

      Those phones likely have a proprietary implementation of TLS/SRTP so I wouldn’t be surprised that it doesn’t work. When those phones were released almost no one had a solution for secure voice…. only years later did Nortel/Avaya have a SRTP solution using their own Call Servers.

      Cheers!

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