Freepbx Install Unistim
Voip Contents Show All: By Date: 1120e keys should be programmed as shown. Example 1120e settings and TNB record, built on a CS1000 virtual loop. Reset a Cisco 2960 (or 2950) password without erasing the configuration. Procedure to reset a Cisco 2960 (or 2950) to factory defaults by deleting the configuration files. 4550T firmware upgrade procedure 4550T firmware upgrade procedure 4550T configuration for closet use 4550T configuration for PBX use Installation notes for Asterisk PBX with OpenVPN on CentOS6 Configure Auto Provisioning using a combination of DHCP, and provisioning file(s) hosted on a TFTP server.
A CS1000 Signaling Server provides access to the SS and CS command line interfaces Basic configuration guide for a Nortel Baystack 470 Ethernet switch Procedure to download the configuration image using Trivial File Transfer Protocol (TFTP). Task overview to integrate a BCM and a CS1000 via IP BCM Destination Codes and Routes are used to send calls to external service or tandem PBX Network Routing Service, or NRS, is responsible for directing network calls to the appropriate end point.
Nortels 12 port BCM style Ethernet switch with Power Over Ethernet (PoE) Make a backup copy of the NRS database Basic interface, routing, dial-peer, and trunk card commands for CISCO VoIP applications. Before you can use a phone with this system it Table of basic specifications for CS1000S, CS1000M, and CS1000E PBX systems CS1000 IP Trunks are built on virtual loops, and require a virtual D channel CS1000 IP Trunks are built on virtual loops, and require a virtual D channel Configuring Win. 2000 DHCP Server for Auto-Configuration of i2004/i2002 IP Phones How to configure PuTTY to connect to a CS1000 (pre Rel.
7.5) using RLOGIN. Use this procedure to enable DHCP for Nortel/Avaya IP Phones using a SR2330. Windows DHCP server configuration, used when deploying Avaya 1120 VoIP sets.
Windows DHCP server configuration, for use in conjunction with with Avaya 1120 VoIP sets. Network Routing Service, or NRS, is responsible for directing network calls to the appropriate end point. This is how you setup a main ERS 4550 that has trunking allowed to a closet ERS 4550 with IP phones.
The status LED indications on a 4500 series Ethernet Routing Switch (ERS). Use this procedure to restore the factory default configuration for Nortel/Avaya data switches. How to connect up to eight 4000 Series switches in a stack to provide up to 400 ports. How to reset IP Phone parameters to the factory defaults. (1120e, 1140, 1230, etc.) How to force install firmware on Avaya CS1000 IP Phone Series. (1120e, 1140, 1230, etc.) The following screen shots show the default G729a codec settings in Element Manager from a 4.5 or 5.5 system. Example i2004 TNB record, built on a CS1000 virtual loop.
Nortel IP Telephones must be configured to work with your system. All the parameters that can be provisioned via the Info-Block provided in a table Loading Signal Servers Procedure to upload and distribute Avaya IP phone firmware from within Element Manager. IP Phone Set Comparision - 1200 vs. 1100 Series How to enter diagnostics mode on a Nortel IP set.
Loading CPPM Sig-Servers How to use the CS 1000 Signaling Server Software Install Tool, and Element Manager, to build Leader and Follower NTDU27 servers and bring into service. DHCP Server Options for Auto-Configuration of i2004/i2002 IP Phones Description and programming notes for configuring an NRS Collaborative Server environment How to create Network Routing Service (NRS) SIP/H323 service endpoints. The NRS is responsible for directing network calls to the appropriate end point.
Network Routing Service, or NRS, is responsible for directing network calls to the appropriate end point. Network Routing Service, or NRS, is responsible for directing network calls to the appropriate end point. Organizational Unique Identifiers, or OUI's, used by Avaya/Nortel IP Phones An example of using hierarchal provision files using system, zone, and type provisioning files How to install Asterisk and FreePBX on the Raspberry Pi, using a Mac with OS X.
Set registration process in an SRG configuration. The IP address, subnet mask, gateway, etc, must be correctly set BEFORE the Shoretel is connected to a network. How to install or upgrade SIP or UNIStim firmware on Avaya CS1000 IP Phone Series. (1120e, 1140, 1230, etc.) How to configure PuTTY for serial communications The Survivable Remote Gateway 50 (SRG50) is a new member of the Nortel family of survivable IP telephony branch office solutions How to route local incoming trunks in an SRG configuration. How to route to local outgoing trunks in an SRG configuration.
A CS1000 Signaling Server provides access to the SS and CS command line interfaces How to build Voice Gateway channels (VGW's), which are used for IP to TDM conversion FAX Configuration Recommendations in the Call Server 1000. Configure Automatic Logon for Windows Server 2008, Windows Vista and Windows 7. Do this to automatically restart applications (which don't run as services) after a power failure. Packet capture utility (sniffer) which can be used to view SIP (and UNISTIM) messages, or diagnose network problems.
Over the past weekend I set out to setup, an open source communication server, to test some of the issues reported in a over on the. I had an Avaya 1120e and 1165e IP phone available to test with, however, both phones were running the software for the Avaya Communications Server 1000. I needed to upgrade them to support.About a year ago I 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 SoftwareYou’ll need to download the SIP software from the Avaya Support website.
You should be able to retreive the SIP software from this 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.binTFTP ServerYou’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 from Philippe Jounin on any Microsoft Windows XP, Vista or Windows 7 personal computer. I download the and exploded the files to D:Temp. TFTP FilesWith 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. FWDOWNLOADMODE AUTOVERSION SIP1165e04.00.04.00FILENAME SIP1165e04.00.04.00.binPROTOCOL TFTPSERVERIP 192.168.1.6SECURITYMODE 0You’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. UpgradeYou 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 writing.SIP1120e04.00.04.00.binVERSION SIP1120e04.00.04.00Once 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. DNSDOMAIN asterisk.homeSIPDOMAIN1 asterisk.homeSERVERIP11 192.168.1.6SERVERPORT11 5060SERVERRETRIES1 3DEFUSER2 ASTERISKVMAIL 5000VMAILDELAY 300DEFLANG EnglishDEFAUDIOQUALITY HighENABLELLDP YESADMINPASSWORD 26567.738ADMINPASSWORDEXPIRY 0# Settings to disable extended licenseMAXLOGINS 1USBHEADSET LOCKEXPMODULEENABLE NOENABLESERVICEPACKAGE NOIMMODE DISABLEDAVAYAAUTOMATICQoS NOVQMONPUBLISH NOSIPTLSPORT 0ENABLEBT NO# Enable SSHSSH YESSSHID adminSSHPWD adminThe settings above disable any advanced features and allow the IP phone to run a basic SIP configuration.Cheers! Unfortunately I had an 1140E SIP phone running in my apartment in Chicago for fun and it died on me after 30 days due to the fact I didn’t have a license token from Avaya.
I was running the phone on the new SIP 4.0 above.I’ve called countless Avaya partners to try and purchase the license for this and an 1120E sip phone I have and out of 20 partners I called, I wasn’t able to get a channel to purchase this keycode. I will admit, all 20 of them didn’t know what I was talking about.
Does anyone have a contact where I can purchase one and how much it is? I can continue running the phones on SIP 2.2, and I will admit, I like the Nortel Splash screen coming up on these phones over the new Avaya one.:)My SIP provider is VOIP.MS by the way.Do you have any advice that can help me legally obtain the SIP license? If I can get this worked out, I’ll go ahead and think about getting an 1165E. One problem with the 1165E, especially if it is used as a receptionist phone with KEM’s, the bluish color on the KEM will not match the color display on the phone. I know Cisco has a similar issue with their 7970’s and they have addressed it by providing a KEM with dark blue background and white letters.Thank you,Joe. Hi Joe,You don’t need to purchase the extended license to enable basic SIP functionality.
You simply need to disable the “advanced” feaures that require an “extended” license key.Have a look at the discussion thread (link above) and that should answer your question. You’ll need to setup either a TFTP server or an HTTP server where the IP phone can download the provisioning files.
You can also configure the IP phone manually through the handset but that can be an adventure.Let us know how you make out!Cheers! Same problem with Nortel 1140e phone with 0625C39 Unistim firmware.Phone stops with message “Upgrades are available”.
DONLOADMODE FORCED does nothing.When i renamed f/w file to “1140e.img” and changed FILENAME parameter in.cfg to same, phone shows “Downloading upgrades” for 2-3 seconds and later changes diaplay to “Connecting to S1”, later “Server unreachable”, “restarting” and so on, as it booting as usual.tftpd64 log shows that phone downloads.cfg and does not try to downloads.imgConnection received from 192.168.0.172 on port 1024 10/10 22:44:29.334Read request for file. Mode octet 10/10 22:44:29.334Using local port 60889 10/10 22:44:29.334: sent 1 blk, 113 bytes in 0 s. 0 blk resent 10/10 22:44:29.349Connection received from 192.168.0.172 on port 1024 10/10 22:45:59.970Read request for file. Mode octet 10/10 22:45:59.970Using local port 61993 10/10 22:45:59.970: sent 1 blk, 103 bytes in 0 s. 0 blk resent 10/10 22:45:59.970. I have an 1140e (NTYS05BCE6 with firmware 0625C6R) that I have been trying to convert to SIP. Have the SIP1140e03.02.16.00.bin file and a 1140e.cfg file in the TFTP server directory for the phone to pick up.
All i get at the TFTP server isConnection received from 192.168.0.240 on port 5000 date & timeUnexpected request 65535 from peer date & timeReturning EBADOP to peer date & timeAnd this just keeps repeating. It is the correct IP for the phone.It doesn’t even seem to be requesting the cfg file.
Is there a setting in the admin menu that perhaps need to be changed. I set up the phone with a fixed IP, did all of the configuration manually but I have yet to see a site that explains (in detail) all of the necessary settings in there prior to a conversion to SIP. Hi Lee,How are you configuring the TFTP server?
What file(s) is the IP phone requesting from the TFTP server? The logs from TFTPD32 should show you exactly what the IP phone is requesting as it boots up. You should see it requesting the file 1140e.cfg which needs to contain all the information referenced above.The process is laid out for you in the above post.
The IP phone will request the cfg file from the TFTP server based on the model, that cfg file will contain the instructions and filename of the firmware that the IP phone will upgrade/downgrade to.Firmware 0625C6R is UNIStim release 3.3 so the ability to upgrade the IP phone should definite be there.Cheers! Thanks Michael. I tried on a SIP 1230 and I does work, it’s now back to UNIStim on my 450 R6.But now I am facing a weird situation, I got a 1120 SIP configured, I thinks this one was on a CS1000 since the version is SIP 02.02.21.00.
Problem is that there is no settings for tftp only provisionning server and I did put my tftp address but no request are being received. When it’s booted I get UserID on the main screen and a configuration menu (pwd color.set) with S1 & S2. I’ve put my BCM info but once again it’s doing nothing.I also tried the factory reset sequence.
Doesn’t work eitherI think I’ll put this phone on eBay?&. Hi MIchael,Thanks for the detailed upgrade info. I have a few 1120e’s, firmware 0624C39. I’ve been unsuccessful in upgrading to firmware SIP1120e04.00.04.00 using your guide. The phone does contact the tftp server, appears to start the download, but rather than going through the reading/writing process I expect, the phone tries to contact S1 and will loop continuously when contact fails (as expected since I’m doing the upgrade on a LAN segment without a PBX). The tftp log shows only 1 block of about 248 bytes transferred, it appears that the cfg file is read but download of the binary never starts. Is my current firmware an issue?
Do I need to upgrade to a later version of the UNISTIM firmware before making the jump to SIP? Hey David and Michael,That is good news that the CS1000/CS2100 will be around for a while. I am quite keen to the Nortel styled features such as ring tones, dial pads, feature buttons and in general any type of legacy Nortel telephone. With the existence of the CS line in our telecom world, users will still have the familiar Nortel phones but with the Avaya name on them instead.As far as BCM goes, I bought myself a backup system for my home in case mine ever goes out. If both go out, I’ll either convert completely to Nortel SIP 11xx phones or get a small CS1000 somehow.a bit much for an 800 sq. Loft but it is the THE telephone systems of all systems in my opinion. I am using SIP trunking with 11xx phones on the BCM as well as a standalone 1140/1120, Avaya 9630 and Cisco 7970 color display on 3CX via VOIP.MS.you can say I am quite the phone geek!
Hi I have a 1140e I followed the commands, got the phone working, but after playing with some settings I stuffed it up and now just continually downloads from the tftp server. I have continued with the instructions on this site and for the life of my I can not get it to downloads users.dat the phone isn’t even requesting it from my server. How can I make it re flash the firmware or make it download the users.dat file, I have tried DOWNLOADMODE AUTO DOWNLOADMODEAUTO DOWNLOADMODE FORCED DOWNLOADMODEFORCED but it just wont request it from the server.
HI there, I am having trouble yet again on trying to load the firmware to my 1140E SIP phone.I am getting the “Could not obtain license” message” and I’ve followed everything you have said here. You could give it a go. Not sure but I done think it will go back to the full factory settings. It should only go back to the current firmware factory settings, if you can fine the old unistim firmware that was on the phone and flash back to that then reflash to the newer SIP firmware it might fix your problem, but for a factory reset type.73639(DeviceMacAddress)##type you mac in numbers for eg. 1A2B3C4D5E6Fwould translate to63to brake it down for you1=1 A=2 2=2 B=2 3=3 C=2 4=4 D=3 5=5 E=3 6=6 F=3so you would type.2435363##and it should then ask you “Are You Sure?” Y or N and then after you confirm it will restart. Let me know how you go.
Did you give the factory reset? How did it go?Actually. In network settings check the status of DHCP check to see if it’s set to Full Partial or Manual, if you know all your network setting try setting it to Manual in input the IP Address of the phone, the Netmask, The Gateway, and the servers (witch are probably already set, but check them just the same) and that should fix your problem, given you will have to manually change the IP settings of the phone if the settings change, but small price to pay for an quick fix. Also is this device the only one have trouble the DHCP, no other computers taking too long for DHCP? Maybe turn the DHCP server off for 30 seconds and then back on. Never know might help.
In network settings check the status of DHCP check to see if it’s set to Full Partial or Manual, if you know all your network setting try setting it to Manual in input the IP Address of the phone, the Netmask, The Gateway, and the servers (witch are probably already set, but check them just the same) and that should fix your problem, given you will have to manually change the IP settings of the phone if the settings change, but small price to pay for an quick fix. Also is this device the only one have trouble the DHCP, no other computers taking too long for DHCP? Maybe turn the DHCP server off for 30 seconds and then back on. Never know might help.
So I am staying in NYC right now, so I won’t be able to get to test the Phones till Sunday. I found out I have free internet at this hotel, I should have brought my 1120e along. I ll try the manual dhcp on Sunday.Btw, I am staying at a Renaissance Marriott and I was hoping for an 1120e In my room like I had at the JW Marriott in grand rapids, no such luck, the hotel has a boring digital Mitel system, not a big fan Of Mitel though. One of the Nortel execs used to be on Marriott s board of directors until 2008, probably exPlains why this 2008 hotel doesn’t have Nortel, dissapointing! I will tell you, I am out here for work for my company Walgreens, and all of our Duane reade stores now have Nortel switches, MiCS I believe!Joe. Well Michael and my fellow SIP Avaya/Nortel Friends,Performing this procedure:The method to factory reset an Avaya 1140 set is:.73639MAC## – where MAC is the MAC address of the phone using the 2 key for A/B/C and the 3 key for D/E/F. You can also think of it as.RENEWMAC##.fixed the problem and my 1120E License Status: Now reads: No Request.does that mean the problem is more than likely fixed?Also, it doesn’t take my phone 5 minutes to reboot and find the DHCP server anymore.I’ll keep you up to date!
Michael,I must also add, can you look over my configuration files above?? They are exactly as copied from your files and I just want you to double check them.Also, What on the phone tells you if the extended feature set has been disabled and the phone won’t shut off after 20 or 60 days? When I RESET to factory default the 1120E and reinstalled the CFG files you created, they worked and the License Status now reads: NO REQUEST. My 1140E hasn’t given me any problems and it is over the evaluation period and says: NO REQUEST. Does that mean I am smooth sailing?Thanks.Joe. Michael,I ordered a Unistim Avaya 1220 phone for $55 on ebay, so I am going to play with making it SIP.How do I get to the TFTP settings on this phone as opposed to the 1140E?
I can uplink my 11xx phones to my TFTP server, but do I do it the same way with the 12xx sets?Can I use the same configuration files for my 11xx phones and just change around the file names to match the 12xx series?I saw something about having to put 12xxboot.cfg for the config. Can you clarify which files I should be using to upload SIP to the phone?When I am asking about getting to the TFTP server on my IP 12xx set, do I just do the 4 key swipe on the softkeys when the phone flashes Avaya?Thank you!Joe. Hi Joey,That’s UNIStim v2.1 firmware release pretty old. I see a problem right away with the files you trying to use. You are running a firmware for the 1120e but you are trying to upgrade to a firmware for the 1150e.
You need to use a firmware that is compatible with the 1120e IP phone. You should be able to download the firmware from the Avaya support website.
Since you are running such an old version you probably need to review the release notes I believe you need to upgrade to at least UNIStim v3.x before you can jump to either 4.x or 5.x.Good Luck! I’ve just started playing with all this VoIP stuff the other day, I’ve built a trixbox and have successfully added an Aastra 480i, 9133i and Nortel 1230 UNISTIM, however I am stuck trying to get an 1120e on the system.
I converted the 1120e to SIP (from UNISTIM) but can’t figure out how to add the phone withing trixbox. With the 1230 I had to manually add to the UNISTIM config file and then create the extension and put in the special dial rule. Is it the same for the SIP? Any help would be greatly appreciated! 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. 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.
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. 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! 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. 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 1.8any suggestions?Thanks! 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. 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 reboot1) 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!
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:-).
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. Hi Mike,i purchased one of this particular Avaya 1140E. I prepared the cfg File according to your Instructions. I use TFTP32 for upload. I configured a DHCP Server on TFTP32in 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 manuallyThanks a lot for Assistance.This is how it looks like.FWDOWNLOADMODE AUTOVERSION SIP1140e04.03.18.00FILENAME SIP1140e04.03.18.00.binPROTOCOL TFTPSERVERIP 192.168.1.100SECURITYMODE 0. Hi MichaelWe 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 Directory2.
Key Expansion Modules3. 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 RegardsBradley. Dear Michael!Finally I could convert my ip phone 1120e to SIP, but I have got some dudes:“Licensing Info” show me:License TokensLicense mode: Node lockedLicense status: No requestToken expiry: 31 day(s)Token type: EvaluationTokens requested: 0Tokens acquired: 0ANDOn “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!
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.
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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.
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Also, I’m wondering if you can post some dial plan example.Thanks,David. I realize this post is really old so don’t know if it’s still active.
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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.