Why the excitement?

E-Metrotel claimed they could support legacy Avaya (formerly Nortel) UNIStim IP phones as well as legacy Norstar digital phones. That statement quickly grabbed my attention knowing that the Norstar is probably the widest deployed small office key system in the United States. I actually still support 5 Norstars in my network because we haven’t found a cost effective replacement for them and in short they just plain work.

I was able to open dialog with Jay and Chris from E-Metrotel who graciously sent me an evaluation UCx50 unit for testing.

Demo Unit

The configuration of the UCx50 was fairly simple and straight forward. It comes out of the box with a default IP address which you can connect to and quickly reconfigure to match your IP addressing scheme. I was quickly able to configure and add 2 SIP extensions for which I used 2 Avaya 1220 IP phones which were running SIP firmware. With that success under my belt I proceeded to attach 2 Avaya i2002 IP phones which were running UNIStim. When configuring the extensions for the UNIStim handsets I selected “Generic Nortel Device” and provided the MAC address of each IP phone. With that all done I quickly had all 4 IP phones working and was placing calls between them without issue so I decided to push the envelope. I setup a digital gateway using the hardware from an BCM/SRG along with a IDE flash drive provided by E-Metrotel. This required me to physically remove the internal hard drive from the BCM/SRG hardware and replace it with an IDE flash drive which was pre-loaded with software by E-Metrotel. As with the UCX-50 itself it was pretty simple to configure the digital gateway and get it up and running. I did spend some time trying to figure out how the digital gateway mapped the 25 pair block I had the Norstar digital phones connected to. Eventually I had calls flowing between the 2 Norstar M7310 phones and ultimately between all the digital and IP phones.

Hardware

The original hardware I tested utilized a Zotac small form factor desktop computer with;

• Intel Atom CPU D525 @ 1.80Ghz
• 2 Gb RAM
• 220 Gb Disk Space
• Ethernet 10/100Mbps

Since that time E-Metrotel has refreshed the hardware  to include the following;

• Intel DN2800MT
• 2GB RAM
• 64Gb SSD Hard Drive
• 6 USB 2.0 ports
• Ethernet LAN 10/100/1000 Mbps
• PCI-e V2.0 Expansion Slot (T1/E1 PRI trunk card)

There are a number of models available depending on the number of IP phones you are looking to support.

•  UCx50E – 80 extensions can be enabled, supports 80 concurrent calls
•  UCx450 – 450 extensions can be enabled, supports 225 concurrent calls
•  UCx1000 – 1000 extensions can be enabled, supports 500 concurrent calls, (redundant power supplies, network i/f’s and Raid 6 Hard drives)
• UCx000 – 2000 extensions can be enabled, supports 800 concurrent calls and up to 15 UCX virtual machines (dual CPU, redundant power supplies, network i/f’s and Raid 6 Hard drives)

The introduction of the UCx2000 is more for service providers looking to support multiple customers on a single piece of hardware.

Dashboard

Software

The UCx50 is based on CentOS Linux 5.9 running Asterisk v1.8 along with FreePBX.

Features

The UCx50 supports the following features;

• SIP Trunks
• SIP Phones/Clients
• Nortel IP and Digital Phones
• Voicemail/Unified Messaging
• Integrated CDR
• Meet Me Conferencing
• Call Recording
• Call Centre & IVR
• Find Me / Follow me
• Full suite business features

The UCX-50 even supports Shared Call Appearances (SCA) which has been a challenge for sometime with Asterisk based solutions. Unfortunately I didn’t have the time to test that feature.

Questions / Interview

Q. Why did you chose Asterisk?
A. We choose Asterisk because it was deemed a mature and robust software structure, it has 95%+ of the features that businesses are looking for and it has thousands of customer installations across every industry segment. Also it already has a vibrant eco-system of active sales/service companies, of contributing code developer/bug fixers and a large and growing number of application/solution providers for almost any business requirement.

Q. Does E-MetroTel contribute back to the Asterisk codebase?
A. Yes, in the past we contributed fixes back to the Asterisk project.  Our product is currently based on the Asterisk version 1.8, which is a LTS release in the maintenance mode – due to that, only important security fixes are accepted for the 1.8 stream.   We are looking into switching to Asterisk 11 in the near future.  Once we complete this task, we’ll have several enhancements that we are planning to contribute to the Asterisk 11/12 stream.

Q. It appears that E-MetroTel is using Elastix/FreePBX for the GUI?
A. Yes. One of the best values in going with an open source platform is that you abandon the NIH mentality and you have a wide variety of options to choose from when you are looking for a particular feature or option. For our management GUI we choose the Elastix/FreePBX as a core starting framework.

Q. Does E-Metrotel contribute back to the Elastix / FreePBX codebase?
A. Our product is currently based on the FreePBX version 2.8 (the same version that is used by Elastix).  This FreePBX version is no longer maintained by the open source community – hence no changes can be contributed.  We have the change to a newer version of FreePBX on our roadmap.  We should be in a position to contribute to the FreePBX open source project once we complete the transition to the latest FreePBX version. We have been contributing fixes to the Elastix open source project.

Q. How is the product licensed beyond the purchase of the base unit?
A. We license in a per extension basis. Extensions can be purchased in blocks of 1, 5, 20 and 50.

Q. How can users find a reseller?
A. By directing such a query to E-MetroTel via our web form, via email (info@emetrotel.com) or calling us at  214-556-5917 .

Q. Have you tested your solution with any SIP based providers? Verizon, XO, AT&T, Nextiva?
A. At this time, we have seven SIP trunk providers listed in our GUI pull-down menu with VoIP providers.  We typically add a new provider to this list whenever we determine that a larger number of customers uses or is planning to use a provider that we haven’t included yet (we register an account with that provider, perform validation testing to determine the proper configuration and then add the provider to the list).

Q. Any recommended SIP providers?
A. We like voip.ms for their QOS and pricing model.

Q. Is there any remote support built into the product?
A. Yes. We provide a built-in VPN and we host a service out of our Plano office which our partners can use to get secure remote access for monitoring and provisioning services. It’s a free service for our gold level partners.  One of the great thing about having such a network-centric product is the ability to leverage all the internet-based desktop sharing apps, or our remote support built-in VPN, for remote troubleshooting and support. All of our products come with 1-year support and warranty included.

My Thoughts

If you are a legacy Nortel customer then the UCx solution could potentially help you migrate to an IP based solution while maintaining your older digital phones or legacy UNIStim IP phones. When upgrading from a legacy key system a large portion of the cost is handsets so this solution could potentially save businesses a lot of money. More importantly it provides the ability to migrate overtime to an all IP solution without the need to forklift the entire telephony environment overnight.

If you are at all interested I would suggest you hit up the folks at @E_Metrotel for more information.

Cheers!

We have a winner! – Avaya Sweepstakes

We had a pretty good response to the sweepstakes we ran over on the discussion forums last month. In the end Hugo from Quebec Canada was our lucky winner. Thanks to everyone for participating and congratulations to Hugo, we hope you enjoy the iPad2 WiFi+3G! Thanks to Avaya for (unofficially) sponsoring this event. It’s exciting to see that there are folks within Avaya that are still very passionate about their products and willing to step up and support both the products and their customers.

Avaya’s New Look

As you may of may not have noticed by now there’s a new look over at the Avaya website and the support website. I’m pretty impressed with the short time I’ve already spent on the support site. The search function is refreshingly quick and fairly thorough. It’s easy to quickly find the product your looking for by browsing through the “Products” menu so by just searching for it.

Unfortunately it looks like they used different field names for the user login, so Firefox couldn’t fill in the field. :(

I generally like it… what do you think?

If your a member of the discussion forums head on over and post your thoughts in this thread. If your not a member why not register, it’s free!

UNIStim Software Release 5.4 for IP Deskphones

Avaya has released UNIStim firmware 5.4 for their IP deskphones;

• 0621C8L for the 2007 IP deskphone
• 0623C8L, 0624C8L, 0625C8L, 0627C8L, 0626C8L for the 1110, 1120E, 1140E, 1150E and 1165E IP deskphones
• 062AC8L for the 1200 series IP deskphones

As always I recommend you review the release notes for all the details.

The following fixes are included in this release;

• wi00966639 External – Intermittent issue where the 1100 Series IP Deskphones fail to open socket resulting in no speech path.
• wi00926866 External – PC authentication with 802.1x (EAPOL) does not work if IP Deskphone re-registers to TPS after a Network failure.
• wi00960325 External – LLDP Time to Live expiration – an error message is displayed, but pressing any key enables the set to work as expected. (SR# 1-2340185912)

I know I’ve personally observed the “LLDP TTL Expired” message on my IP phones and I recall quite a few comments from a number of readers.

Cheers!

The installation was pretty straight forward, however, there were a few small issues that I had to deal with. Initially I was unable to connect to the server and found that the firewall was enabled, so I had to disable the firewall with the following commands, service iptables stop, chkconfig iptables off. I was also getting a weird error in the FreePBX gui when I tried to apply the configuration;

exit: 126
sh: /var/lib/asterisk/bin/retrieve_conf: Permission denied

…this turned out to be an issue with SELINUX, so I had to edit /etc/selinux/config and disable SELINUX (a reboot is required for the change to take effect). Once I did those few steps I was ready to create some extensions so I created 1001 and 1002 and set their password (secret) to ‘abc123’.

The Avaya (legacy Nortel) IP phones can be provisioned from a TFTP server so I installed a TFTP server on my Asterisk server using yum install tftp-server. Then I enabled the TFTP server with chkconfig tftp on and finally I had to restart xinetd with service xinetd restart. I placed the files I needed in the /tftpboot directory including 1220SIP.cfg, 1120eSIP.cfg and users.dat (these filenames are case sensitive on a Linux server – if you use a Windows server such as TFTPD32 then the case is not an issue). I configured my local DHCP server to offer DHCP option 66 (TFTP Server) and I was off and running. The 1220 and 1120e both booted, download the provisioning files from the TFTP server, and connected to the Asterisk server. I entered the username and passwords and I was logged in and running in seconds placing calls between the two handsets.

I had to refer to my original post on the forums on what settings I needed to disable the extended license;

http://forums.networkinfrastructure.info/nortel-ip-telephony/disabling-features-from-extended-feature-set-on-ip-deskphone/

Here’s what the configuration files on the TFTP server looked liked, the 1220SIP.cfg file contained the following lines;

[FW]
DOWNLOAD_MODE AUTO
VERSION SIP12x004.03.09.00
FILENAME SIP12x004.03.09.00.bin
PROTOCOL TFTP

[DEVICE_CONFIG]
DOWNLOAD_MODE FORCED
VERSION 000200
FILENAME users.dat

[DIALING_PLAN]

The 1120eSIP.cfg file contained the following lines;

[FW]
DOWNLOAD_MODE AUTO
VERSION SIP1120e04.00.04.00
FILENAME SIP1120e04.00.04.00.bin
PROTOCOL TFTP

[DEVICE_CONFIG]
DOWNLOAD_MODE FORCED
VERSION 000200
FILENAME users.dat

[DIALING_PLAN]

The users.dat file contained the following lines;

DNS_DOMAIN local
SIP_DOMAIN1 asterisk.local
SERVER_IP1_1 192.168.1.10
SERVER_PORT1_1 5060
SERVER_RETRIES1 3

VMAIL 5000
VMAIL_DELAY 300

DEF_LANG English
DEF_AUDIO_QUALITY High

ADMIN_PASSWORD 26567*738
SSH YES
SSHID admin
SSHPWD admin
# 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

I did have to re-configured the 1220 to AllAut before it would honor the settings in the TFTP provisioning file.

Cheers!

This software release is compatible with the following Call Server platforms;

• Avaya IP Office R8.0 (1220, 1230, 1120E, 1140E IP Deskphones only)
• Avaya CS1000 R7.0 and 7.5
• Avaya CS2100 SE13
• Avaya Aura® Communications Manager 6.0
• Avaya Aura® Session Manager 6.0

I’ll refer you to the release notes for all the details.

Cheers!

• 0621C8J for the 2007 IP deskphone
• 0623C8J, 0624C8J, 0625C8J, 0627C8J, 0626C8J for the 1110, 1120E, 1140E, 1150E and 1165E IP deskphones
• 062AC8J for the 1200 series IP deskphones

As always I recommend you review the release notes for all the details.

This release includes the following enhancements;

• LLDP Advertisement on PC port
• Support for new audio profile configuration option through Zero Touch Provisioning to support Australian/New Zealand S004 audio standard (applies to 1165E only)
• Support for BCM to IP Office migrations to upgrade IP Deskphones from UNIStim to SIP (1120E, 1140E, 1220, 1230 IP Deskphones only)

I noticed the following warning concerning Avaya’s (formerly Nortel) Contact Recording and Quality Monitoring (CRQM) if you are using Secure Calling;

IMPORTANT NOTE: If a customer is using secure call recording with Avaya Call Recorder (ACR) then they should not upgrade to UNIStim 5.3 until an ACR patch (101055) is available and the functionality is enabled by updating a specific entry in the ACR 10.1 properties file. Note the 101055 patch for ACR10.1 is Generally Available, however, the default operation with this patch applied is to NOT enable the new functionality. Therefore if a customer site is using UNIStim 5.3 and secure call recording, then a specific entry needs to be added into the ACR 10.1 properties file which enables the new functionality. The specific string that needs to be entered will be published once system level regression testing is completed, expected later this month. At that time, this Product Bulletin will be updated.

Cheers!

• 0621C8G for the 2007 IP deskphone
• 0623C8G, 0624C8G, 0625C8G, 0627C8G, 0626C8G for the 1110, 1120E, 1140E, 1150E and 1165E IP deskphones
• 062AC8G for the 1200 series IP deskphones

As always I recommend you review the release notes for all the details.

Cheers!

This release adds support for the following two new features;

‘More’ Feature Key (IP Office)

SIP Software Release 4.1 for 11xx/12xx IP Deskphones introduced a mechanism for IP Office to configure extra features using the soft keys on the deskphone. Soft keys are the 4 buttons located below the display screen on the phone. Soft key button labels are displayed on the phone display right above the physical buttons, with each label corresponds to the physical button. SIP Software Release 4.1 Service Pack for 11xx/12xx IP Deskphones extends this feature by the addition of a ‘More’ key as the right-most button, indicating more selections are available.

Configurable LLDP Timeout

SIP Software Release 4.1 for 11xx/12xx IP Deskphones provides a mechanism to customize the time to wait for LLDP data from the network.

On reset/power-up, the Deskphone will try to obtain LLDP data (i.e. the VLAN ID) only once at startup. If no response is received from the network switch the deskphone will continue to boot. After that the phone will not retry to get LLDP data until the next reboot. In some cases, the network switch may take too much time to start up. In this case, if the switch applies power to its ports (POE) before the switch is ready to accept network packets from devices (the deskphone) connected to the switch, the LLDP negotiation may timeout before the switch is ready to respond.

Prior to SIP Software Release 4.1 Service Pack, following a reboot (and the ENABLE_LLDP flag is set), the deskphone would wait for LLDP data from switch for up to 30 seconds. If no data was received, the deskphone starts up using the previous data stored in EEPROM.

With SIP Software Release 4.1 Service Pack, a new provisioning parameter has been added to the device configuration file to specify the time to wait for LLDP data from the network switch:

LLDP_WAITING_TIME

Minimum value is 30 seconds

Maximum value is 300 seconds (5 minutes).

I haven’t personally seen the issue that Avaya is attempting to resolve regarding the LLDP timer when connecting the 1100 or 1200 series IP phones to the Avaya Ethernet Routing Switch 5520. I have observed that the switch appears to delay enabling PoE on the ports until it’s far enough along through the boot-up process.

Cheers!

• 0621C8D for the 2007 IP deskphone
• 0623C8D, 0624C8D, 0625C8D, 0627C8D, 0626C8D for the 1110, 1120E, 1140E, 1150E and 1165E IP deskphones
• 062AC8D for the 1200 series IP deskphones

This release of firmware adds support for the following features;

• Support for a Wireless Markup Language (WML) browser on the 2007 IP Deskphone.

UNIStim software release 5.1 delivers quality improvements for all IP Deskphones including software fixes for over one hundred work items, and resolution of seven customer reported cases. A previously reported Product Advisement related to the 1165E IP Deskphone locking up if invalid UserID and password for WML authentication are provided has been addressed in this UNIStim 5.1 software release.

UNIStim Software Release 5.1 is supported on the Avaya 2007, 1110, 1120E, 1140E, 1150E, 1165E, 1210, 1220, and 1230 IP Deskphones only. For details on specific hardware vintages which are supported, please refer to the ReadMe file.

UNIStim Software Release 5.1 is compatible with the following Call Servers:

•     Avaya CS 1000 Release 5.0, 5.5, 6.0, 7.0, 7.5
•     Avaya SRG 50 Release 6.0
•     Avaya SRG 200/400 Release 1.5
•     Avaya CS 2100
•     Avaya BCM 50/450 Release 6.0 (refer to ReadMe for additional information)
•     Avaya BCM 200/400 Release 4.0 (refer to ReadMe for additioanl information)
•     Avaya CS 2100 CICM Release 10.1 MR2, Release 11 MR2

Product Advisement for customers who previously upgraded to UNIStim 5.0 using the workaround related to Zero Touch Provisioning:

For customers who previously upgraded to UNIStim 5.0 and used the recommended workaround detailed in the UNIStim 5.0 Readme Product Bulletin, related to pre-configured REG entries, this workaround needs to be removed to successfully upgrade to UNIStim 5.1. That is, customers who have pre-configured REG entries, (which includes the MAC address, the TN, and Node) within the provisioning file to enable Zero Touch for the IP Deskphones, were previously advised of a problem that could occur related to parsing of the REG entry which prevented the IP Deskphones from coming up as expected. The recommended workaround for customers upgrading to UNIStim 5.0 was to add a comma before the semi-colon of the REG entry within the provisioning file. As noted in the UNIStim 5.0 Readme Product Bulletin, this issue is now fully addressed in this UNIStim 5.1 software maintenance release, and customers who applied the workaround will need to remove the comma to successfully upgrade to this UNIStim 5.1 software release.

You should refer to the release notes for all the details.

Cheers!

• (101102-26211, wi00838178, wi00828501)
  Corrects an issue running on Windows 7 where the IP Softphone 2050 does not connect to the call server after displaying “Terminal Manager Connect”.
• (100721-77062, wi00832321)
  Corrects an issue where the IP Softphone 2050 may hang if it receives a call after being idle for several hours and the call is answered immediately.
• (wi00839244)
  Corrects an issue where the IP Softphone 2050 would crash when clicking “Add Link To” in the Local Directory.
• (wi00842584)
  Resolves an issue with duplicate audio streams where the Receive audio stream is not reopened after both the Rx and Tx streams have been closed.

The release notes can be found here. You can also find additional information in the IP Phone Fundamentals manual which is here.

Cheers!

• 0621C8A for the 2007 IP deskphone
• 0623C8A, 0624C8A, 0625C8A, 0627C8A, 0626C8A for the 1110, 1120E, 1140E, 1150E and 1165E IP deskphones
• 062AC8A for the 1200 series IP deskphones

This release of firmware adds support for the following features;

• Support for wideband (G.722) codecs on 1120E, 1140E, 1150E, 1165E, 1220, and 1230 IP Deskphones.
• Support for Avaya Notification Solution (ANS) and Avaya Push API which provides the ability to push text, graphics, and audio messages to 1100 Series, 1200 Series and the 2007 IP Deskphones.
• Support for a Wireless Markup Language (WML) browser for 1140E, 1150E, and 1165E IP Deskphones.

Product Advisement for UNIStim 5.0 related to Zero Touch Provisioning:

For customers who have pre-configured the REG entries (which includes the MAC address, the TN, and Node) within the provisioning file to enable Zero Touch for the IP Deskphones, be advised that a problem can occur related to parsing of the REG entry that may result in the IP Deskphones not coming up as expected, and instead continuing to reboot when UNIStim 5.0 is loaded onto the units. The issue will impact IP Deskphones that have been pre-configured for Zero Touch and where the REG lines still exist in the REG entry. This is a known issue that will impact IP Deskphones that have a matching MAC address already pre-configured in the REG entry. This issue will be fully addressed in the upcoming UNIStim 5.1 maintenance release expected in March 2011, and customers may want to delay updating IP Deskphones and Call Servers until that time due to this issue.

In the interim, to avoid this issue the following workaround is recommended: Before loading UNIStim 5.0 onto the IP Deskphones, customers are advised to add a comma before the semi-colon of the REG entry. If UNIStim 5.0 is already loaded, and the issue exists (where the phones continue to reboot instead of coming up), the customer can then add a comma as specified above and the IP Deskphones will come up.

Example: reg= 0021e1ff59cb cs1k s1 3380 096 00 00 18,;

I also recently had a discussion with another telephony expert around the issues of running a PC at 100Mbps (connected to the IP phone) when the 1100 series IP phone is connected to the network via 1Gbps or vice-versa. Here’s the relevant blurb from the release notes that details the problem.

Throughput may be slow for large file transfers on conversions from GigE to 100Mbit (applies to the 1120E, 1140E, 1150E and 1165E IP Deskphones)

In networks in which a PC is connected to the IP Deskphone’s PC port and the PC’s NIC speed is 100Mbit but the network speed is at GigE, large file transfers to the PC can take quite a long time. This is an issue with large file transfers only. Due to the speed mismatch between the phone’s two ports the buffers in the phone can overflow resulting in retransmissions. Although the IP Deskphones support Ethernet flow control (802.3x), the support is only implemented on the phone’s PC port, not on the phone’s network port. Ethernet flow control is a mechanism were the IP Deskphone can request a brief “pause” from the transmitting Ethernet device if the IP Deskphone buffers are about to overflow.
Ethernet flow control cannot be implemented on the phone’s network port, since it impacts the phone’s voice quality. As a result, in environments were the network is GigE but the PC NIC is only 100Mbit, large file transfers from the network to the PC can take quite a long time. On the other hand, since Ethernet flow control is implemented on the phone’s PC port, in environments were the PC NIC is GigE but the network is only 100Mbits, large file transfers should be well managed by the phone’s Ethernet flow control mechanism.

You can find the release notes on Avaya’s website along with the actual firmware/software for the IP phones.

Cheers!

If time allows I hope to post an update in the very near future covering the new Nortel-i2004-B option string. While working with the SIP 4.0 software release in the past few articles I did learn that the Avaya SIP IP phones can utilize a new DHCP vendor class, Nortel-SIP-Phone-A which can allow you a little flexibility when configuring them via DHCP and TFTP.

In the same file below I setup four DHCP scopes; one for 2111/2212/6020/6040 wireless handsets, one for i2002/i2004/1100/1200 series IP phones, one for 1100/1200 series IP phones running SIP and one for all other devices (laptops, desktops, etc). Just a quick note about the example below, you’ll notice that I have no pools in the 192.168.1.0/24 network. All the pools are in the 192.168.25.0/24 network.

#
# DHCP Server Configuration file.
#   see /usr/share/doc/dhcp*/dhcpd.conf.sample
#
# Sample dhcpd.conf file for Avaya (legacy Nortel) IP Phones
#
# Notes: example dhcpd.conf file to illustrate how to configure Avaya
# IP Phones with specific DHCP options for 2000/1100/1200 series IP
# Phones and the 2200/6100 series Wireless IP Phones.
#
# *** WARNING *** WARNING *** WARNING *** WARNING ** WARNING ***
#
# This is just an sample file with specific IP information. You'll
# need to customize this file to your specific IP address scheme
# before you can use it in your environment.
#
# *** WARNING *** WARNING *** WARNING *** WARNING ** WARNING ***
#

ddns-update-style none;
not authoritative;

option nortel-callserver code 128 = string;
option nortel-2245 code 151 = ip-address;
option tftp-server-name "192.168.1.20";

# Vendor Class for i2002/i2004/1120e/1140e/1150e Internet Telephones
class "Nortel-i2004-A" {
  match if substring (option vendor-class-identifier, 0, 14) = "Nortel-i2004-A";
    option nortel-callserver "Nortel-i2004-A,192.168.200.2:4100,1,5;192.168.200.2:4100,1,5.";
    option vendor-class-identifier "Nortel-i2004-A";
}

# Vendor Class for 2210/2211 Wireless Phones
class "Nortel-221x-A" {
  match if substring(option vendor-class-identifier, 0, 13) = "Nortel-221x-A";
    option nortel-callserver "Nortel-i2004-A,192.168.200.2:4100,1,5:192.168.200.2:4100,1,5.";
    option nortel-2245 192.168.99.10;
    option vendor-class-identifier "Nortel-221x-A";
}

# Vendor Class for Avaya 1100/1200 IP SIP Phones (SIP firmware loaded)
class "Nortel-SIP-Phone-A" {
  match if substring(option vendor-class-identifier, 0, 18) = "Nortel-SIP-Phone-A";
    option vendor-class-identifier "Nortel-SIP-Phone-A";
}

# Network Definition
shared-network "mynetwork" {
   subnet 192.168.1.0 netmask 255.255.255.0 {
   option subnet-mask 255.255.255.0;
   option routers 192.168.1.1;
   option domain-name "home";
   option domain-name-servers 192.168.0.1;
   next-server 192.168.1.20;
   default-lease-time 28800;
   max-lease-time 86400;
   }
}

# Network Definition 192.168.25.0/24
shared-network "192-168-25-0" {
   subnet 192.168.25.0 netmask 255.255.255.0 {
   option subnet-mask 255.255.255.0;
   option routers 192.168.25.1;
   option domain-name "home";
   option domain-name-servers 192.168.1.1;
   next-server 192.168.1.20;
   default-lease-time 28800;
   max-lease-time 86400;

   # IP Address Pool for generic devices
   pool {
      range 192.168.25.50 192.168.25.100;
      deny members of "Nortel-i2004-A";
      deny members of "Nortel-221x-A";
      deny members of "Nortel-SIP-Phone-A";
   }

   # IP Address Pool for i2002/i2004/1120e/1140e/1150e
   pool {
      range 192.168.25.150 192.168.25.175;
      allow members of "Nortel-i2004-A";
      deny members of "Nortel-221x-A";
      deny members of "Nortel-SIP-Phone-A";
   }

   # IP Address Pool for 2210/2211
   pool {
      range 192.168.25.176 192.168.25.199;
      allow members of "Nortel-221x-A";
      deny members of "Nortel-i2004-A";
      deny members of "Nortel-SIP-Phone-A";
      }

   # IP Address Pool for Avaya 1100/1200 IP SIP Phones
   pool {
      range 192.168.25.200 192.168.25.224;
      allow members of "Nortel-SIP-Phone-A";
      deny members of "Nortel-i2004-A";
      deny members of "Nortel-221x-A";
      }

 }
}

Cheers!

[root@centos ~]# ssh -l admin 192.168.100.99
admin@192.168.100.99's password:

Welcome to Avaya problem determination tool.

You are connected to IP Phone 1220.
HW version: 1800247F9984E9662A
FW version 04.00.04.00
MAC Address = 00247F99FFFF
IP = 192.168.100.99
Type "bye" to exit current shell.
PDT> prtcfg
*********************SYSTEM CONFIG************************
*** SIPdomain1 asterisk.home
***   defUser user1
***   S1 IP: 192.168.1.6
***   S1 Ports: (UDP:5060, TCP:0, TLS:0)
***   S1 Proto: 2
***   S2 IP: 0.0.0.0
***   S2 Ports: (UDP:5060, TCP:0, TLS:0)
***   S2 Proto: 2
*** CONFERENCE_URI: conference@avaya.com
*** ADHOC_ENABLED: 0
*** MAX_ADHOC_PORTS: 0
*** SIPdomain2
***   defUser user2
***   S1 IP: 0.0.0.0
***   S1 Ports: (UDP:5060, TCP:0, TLS:0)
***   S1 Proto: 2
***   S2 IP: 0.0.0.0
***   S2 Ports: (UDP:5060, TCP:0, TLS:0)
***   S2 Proto: 2
*** CONFERENCE_URI: conference@avaya.com
*** ADHOC_ENABLED: 0
*** MAX_ADHOC_PORTS: 0
*** SIPdomain3
***   defUser user3
***   S1 IP: 0.0.0.0
***   S1 Ports: (UDP:5060, TCP:0, TLS:0)
***   S1 Proto: 2
***   S2 IP: 0.0.0.0
***   S2 Ports: (UDP:5060, TCP:0, TLS:0)
***   S2 Proto: 2
*** CONFERENCE_URI: conference@avaya.com
*** ADHOC_ENABLED: 0
*** MAX_ADHOC_PORTS: 0
*** SIPdomain4
***   defUser user4
***   S1 IP: 0.0.0.0
***   S1 Ports: (UDP:5060, TCP:0, TLS:0)
***   S1 Proto: 2
***   S2 IP: 0.0.0.0
***   S2 Ports: (UDP:5060, TCP:0, TLS:0)
***   S2 Proto: 2
*** CONFERENCE_URI: conference@avaya.com
*** ADHOC_ENABLED: 0
*** MAX_ADHOC_PORTS: 0
*** SIPdomain5
***   defUser user5
***   S1 IP: 0.0.0.0
***   S1 Ports: (UDP:5060, TCP:0, TLS:0)
***   S1 Proto: 2
***   S2 IP: 0.0.0.0
***   S2 Ports: (UDP:5060, TCP:0, TLS:0)
***   S2 Proto: 2
*** CONFERENCE_URI: conference@avaya.com
*** ADHOC_ENABLED: 0
*** MAX_ADHOC_PORTS: 0
*** DNSdomain asterisk.home
*** Config version 000005
*** User Config version 000001
*** Language version 000001
*** Image version 000001
*** Tone version 000001
*** Licensing file version 000001
*** User keys version 000001
*** CTL file version 000001
*** Boot version 000001
*** Security Policy version 000001
*** CRL file version 000001
*** Nortel Key version 000001
*** Misc version 000001
*** Service Pack version 000001
*** SIP_PING 1
*** VMAIL 5000
*** VMAIL_DELAY 300
*** BANNER Avaya SIP Client
*** FORCE_BANNER 0
*** DST_ENABLED 1
*** TIMEZONE_OFFSET 0
*** MAX_INBOX_ENTRIES 100
*** MAX_OUTBOX_ENTRIES 100
*** MAX_REJECTREASONS 20
*** MAX_CALLSUBJECT 20
*** MAX_PRESENCENOTE 20
*** DEF_LANG English
*** DSCP_CONTROL 0
*** 802.1P_CONTROL -1
*** DSCP_MEDIA 0
*** 802.1P_MEDIA -1
*** DSCP_DATA -1
*** 802.1P_DATA -1
*** LOG_LEVEL 255
*** RECOVERY_LEVEL 2
*** AUTO_UPDATE 0
*** AUTO_UPDATE_TIME 0
*** AUTO_UPDATE_TIME_RANGE 1
*** DOS_PACKET_RATE 5
*** DOS_MAX_LIMIT 100
*** DOS_LOCK_TIME 20
*** DEF_AUDIO_QUALITY High
*** DEF_DISPLAY_IM NO
*** MAX_IM_ENTRIES 999
*** MAX_ADDR_BOOK_ENTRIES 100
*** ADDR_BOOK_MODE NETWORK
*** IM_MODE DISABLED
*** ADMIN_PASSWORD 26567*738
*** ADMIN_PASSWORD Expiry Time 0
*** ENABLE_LOCAL_ADMIN_UI 1
*** HASHED_ADMIN_PASSWORD 0
*** HASH_ALGORITHM SHA1
*** HOLD_TYPE rfc3261
*** AUTH_METHOD AUTH
*** ENABLE_3WAY_CALL 1
*** DISABLE_PRIVACY_UI 0
*** DIALTONE
*** RINGINGTONE
*** BUSYTONE
*** FASTBUSYTONE
*** CONGESTIONTONE
*** DISTINCTIVE_RINGING 1
*** CALL_WAITING SPEAKER
*** PCPORT_ENABLE 1
*** LLDP_ENABLE 0
*** BLUE TOOTH Disable
*** NAT_SIGNALLING NONE
*** NAT_MEDIA NONE
*** NAT_TYPE NONE
*** NAT_TTL 120
*** STUN_SERVER_IP1 0.0.0.0
*** STUN_SERVER_IP2 0.0.0.0
*** STUN_SERVER_PORT1 3478
*** STUN_SERVER_PORT2 3478
*** USE_RPORT 0

*** VQMON PUBLISH ADDRESS: 0.0.0.0
***   PUBLISH ENABLE: 0
***   LISTENING R ENABLE [0:= No, 1:=Yes]: 0
***   LISTENING R WARNING: 0 [0:= 80]
***   LISTENING R EXECSSIVE: 0 [0:= 70]
***   PACKET LOSS ENABLE [0:= No, 1:=Yes]: 0
***   PACKET LOSS WARNING : 0 [0:= 1]
***   PACKET LOSS EXECSSIVE: 0 [0:= 5]
***   DELAY ENABLE [0:= No, 1:=Yes]: 0
***   DELAY WARNING: 0 [0:= 300]
***   DELAY EXECSSIVE: 0 [0:= 500]
***   JITTER ENABLE [0:= No, 1:=Yes]: 0
***   JITTER WARNING: 0 [0:= 150]
***   JITTER EXECSSIVE: 0 [0:= 500]
***   SESSION_RPT_EN: 0
***   SESSION_RPT_INT: 60
***   VQMON CONFIG BLOCK *** END ***
***
*** TRANSFER_TYPE: rfc3261
*** ENABLE_PRACK: 0
*** ENABLE_UPDATE: 1
*** PROXY_CHECKING: 1
*** REDIRECT_TYPE: MCS
*** MADN_PRIVACY:
*** MADN_TIMER: 1800
*** MADN_DIALOG: 0
*** IM_NOTIFY: 1
*** DISABLE_OCT_ENDDIAL: 0
*** FORCE_OCT_ENDDIAL: 0
*** DISPLAY_CALL_SNDR_IM_KEY: 1
*** FORCE_CFWD_NOTIFY: 0
*** DEFAULT_CFWD_NOTIFY: 0
*** FORCE_TIME_ZONE: 0
*** SNTP_SERVER:
*** SNTP_ENABLE: 0
*** RTP_MIN_PORT: 50000
*** RTP_MAX_PORT: 50100
*** TOVM_SOFTKEY_ENABLE: 0
*** TOVM_VOICEMAIL_ALIAS: transfertovm
*** TOVM_VOICEMAIL_PARAM: mbid
*** AUTOLOGIN_ENABLE: 1
*** SCA_BROADWORKS: 0
*** SCA_LINE_SEIZE_EXPIRES: 15
*** SCA_HOLD_BEHAVIOR: PUBLIC
*** SCA_APPEARANCES: 12
*** MAX_RING_TIME: 0
*** EXP_MODULE_ENABLE: 0
*** PROMPT_ON_LOCATION_OTHER: 0
*** ENABLE_ANSWER_MODE: 0
*** ANSWER_MODE_MAXALLOWADDR: 100
*** ANSWER_MODE_MICMUTE: 0
*** AUDIO_CODEC1:
*** AUDIO_CODEC2:
*** AUDIO_CODEC3:
*** AUDIO_CODEC4:
*** AUDIO_CODEC5:
*** AUDIO_CODEC6:
*** AUDIO_CODEC7:
*** AUDIO_CODEC8:
*** AUDIO_CODEC9:
*** AUDIO_CODEC10:
*** AUDIO_CODEC11:
*** AUDIO_CODEC12:
*** AUDIO_CODEC13:
*** AUDIO_CODEC14:
*** AUDIO_CODEC15:
*** G729_ENABLE_ANNEXB: 0
*** G723_ENABLE_ANNEXA: 0
*** LOGOUT_WITHOUT_PASSWORD: 0
*** ENABLE_SERVICE_PACKAGE: 0
*** SECURE_INCALL_DIGITS: 0
*** AVAYA_AUTOMATIC_QOS: 0
*** REMOTE_CHECK_FOR_UPDATE: 0
*** INTERCOM_PAGING: 0
*** ALPHA_ORDER_LOC_LIST: 1
*** MAX_LOGINS: 1
*** AUTOCLEAR_NEWCALL_MSG: 0
*** E911_USERNAME: anonymous
*** E911_PROXY   :
*** E911_PASSWORD: 123456
*** E911_TXLOC   : INVITE
*** FM_PROFILES_ENABLE: 1
*** FM_LANGS_ENABLE: 1
*** FM_SOUNDS_ENABLE: 1
*** FM_IMAGES_ ENABLE: 1
*** FM_CERTS_ENABLE: 0
*** FM_CONFIG_ ENABLE: 0
*** FM_LOGS_ENABLE: 1
*** PORT_MIRROR_ENABLE: 0
*** LOGSIP_ENABLE: 0
*** MEMCHECK_PERIOD: 86400 secs
*** SIP_UDP_PORT: 5060
*** SIP_TCP_PORT: 5060
*** SIP_TLS_PORT: 0
*** KEEP_ALIVE_TYPE: OS
*** CONN_KEEP_ALIVE: 120
*** REGISTER_RETRY_TIME: 30
*** REGISTER_RETRY_MAXTIME: 1800
*** SECURE_UI_ENABLE: 0
*** LOGIN_NOTIFY: OFF
*** LOGIN_NOTIFY_TIME: 0
*** ENABLE_USB_PORT: Yes
*** USB_MOUSE: UNLOCK
*** USB_KEYBOARD: UNLOCK
*** USB_HEADSET: LOCK
*** USB_MEMORY_STICK: UNLOCK
*** USB_LOCK_OVERRIDE: No
*** ATA_REGION: NA
*** IPV6_ENABLE: 0
*** PREFER_IPV6: 0
*** IPV6_STATELESS: 1
*** SECONDARY_LOGOUT_ENABLE: 0
*** SRTP_ENABLED 0
*** SRTP_MODE BE-2MLines
*** SRTP_CIPHER_1 AES_CM_128_HMAC_SHA1_80
*** SRTP_CIPHER_2 AES_CM_128_HMAC_SHA1_32
*** SSH 1
*** SFTP 0
*** SSHID admin
*** SSHPWD ****
EAPConfigRead - migrating EAP Configuration data from TFFS
Failed to open file /flash0/EAPDATA.DAT
*** EAP DISABLED
*** EAPID1
*** EAPID2
*** EAPPWD ****
*** CA
*** CA_DOMAIN
*** HOST_NAME
*** SFTP_READ_PATTERNS .cfg,.dat
*** SFTP_WRITE_PATTERNS .cfg,.dat
*** DSCP_OAM: 18
*** DSCP_MEDIA_FLASHOVERRIDE: 41
*** DSCP_MEDIA_FLASH: 42
*** DSCP_MEDIA_IMMEDIATE: 44
*** DSCP_MEDIA_PRIORITY: 45
*** SESSION_TIMER_ENABLE: 1
*** SESSION_TIMER_DEFAULT_SE: 1800
*** SESSION_TIMER_MIN_SE: 1800
*** SET_REQ_REFRESHER: 0
*** SET_RESP_REFRESHER: 2
*** HOTLINE_ENABLE: 0
*** HOTLINE_URL: hotline
*** DoD_ENABLE: 0
*** MLPP_NETWORK_DOMAIN: DSN
*** MLPP_PRECEDENCE_DOMAIN: 000000
*** CALL_WAITING_TONE: 0
*** MAX_APEARANCE: 10
*** DISABLE_SPKRPHN: 0
*** CALL_ORIGIN_BUSY: 0
*** SLOW_START_200OK: 0
*** SPEEDLIST_KEY_INDEX: 0
*** SPEEDLIST_LABEL: SDL
*** SCRNSVR_ENABLE: 1
*** SCRNSVR_UNPRTCTD_ENABLE: 0
*** SCRNSVR_UPASS_ENABLE: 0
*** SCRNSVR_MODE: 0
*** SCRNSVR_DELAY: 10
*** SCRNSVR_TEXT: Screensaver active
*** SCRNSVR_IMAGE:
*** MENU_AUTO_BACKOUT: 30
*** LOGIN_BANNER_ENABLE: 0
*** BLF_ENABLE: 0
*** BLF_RESOURCE_LIST_URI:
*** BG_IMAGE_ENABLE: 1
*** BG_IMG_SELECT_ENABLE: 1
*** USE_BG_IMAGE:
*** USER_FILE_ENABLE: 0
*** USER_FILE_PATH: /
*** DEFAULT_ADDRESSBOOK_FILE:
*** DEFAULT_SPEEDDIALLIST_FILE:
*** DEFAULT_CUSTOMKEYS_FILE:
*** TECH_SUPPORT_LABEL:
*** TECH_SUPPORT_ADDRESS:
*** SERVICE_PACKAGE_PROTOCOL: HTTP
*** SELECT_LAST_INCOMING 0
*** MKI_ENABLE: 0
*** ALLOW_EMERGENCY_PRIORITY_HEADER: 0
*** CALLINFO_IMAGE_ENABLE 0
*** maskSectionDwnloaded 0x0
*** SURV_SIP_SVR_ENABLE: 0
*** REG_REFRESH_TIMER:  86400
*** OUTLINEFONT_ENABLE: 1
*** FONTSMOOTH_ENABLE: 0
*** FIPS_MODE: 0
*** LOGINALPHA_ENABLE: 0
*** PROMPT_AUTHNAME_ENABLE: 0
*** KEEPALIVE_RETRIES 3
*** IP_OFFICE_ENABLE 0
*** USE_PUBLISH_FOR_PRESENCE 0
*** FAIL_BACK_TO_PRIMARY 0
*** CONTACT_HDR_PORT_CS1K 0
*********************************************************

I should point out that this command is available for the 1100 and 1200 series IP phones.

Cheers!

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!

Here are some of the enhancements made in the new software release;

• Improved Licensing
• SIP Support for 1220,1230 and 1165E IP Deskphones
• Shared Call Appearances – CS1000
• IPv6 Support
• SRTP Media Security
• TLS Signaling Security
• Certificate-based Authentication
• Enhanced Screensavers
• Background images
• Support for Avaya Aura Communication Manager / Session Manager

I was having a discussion with “Mike” in the comments section of any earlier post entitled, SIP Software Release 3.0 for IP Deskphones, in which he pointed out some of the issues with the new licensing model. Well it looks like Avaya was paying attention to that thread and made some changes to the licensing that should satisfy the majority of users. (I’m just going to quote directly from the readme.)

Improved Licensing

Licensing was introduced in the SIP 3.0 release. With SIP 3.2, the following changes are made to the licensing mechanism:

• The Standard feature set is now available on all desksets without a token. This provides a basic set of SIP features conforming to RFC 3261 (SIPPING 19) at no additional cost.
• Now, when the phone is registered to a recognized Avaya call server (Avaya AuraTM, AS 5300, CS1000 or CS2100), the Extended feature set is available as well without a token.
• The Advanced feature set is reserved for Federal and DoD features on the AS 5300 call server only
• The feature packages have been re-organized
  • Wideband is part of Standard feature set
  • IPv6 and Broadworks SCA are part of Extended feature set
  • Security is now part of the Extended feature set

If you connect your IP deskphone to a Avaya Call Server (Avaya AuraTM, AS 5300, CS1000 or CS2100), you’ll get all the standard features you would get with the UNIStim firmware. The licensing really only comes into play if you decide to connect your Avaya IP deskphone to a third party call server or SIP provider.

Please make sure to review the product bulletin and the readme for all the details.

Cheers!

Cheers!

I personally used Paetec a few years ago as a local CLEC where they had been providing local and long distance toll access for a number of our facilities over traditional T1/PRI access lines.

Cheers!

Several enhancements have been included in SIP Release 3.0 for the 1100 series phones including User Interface and Preferences enhancements, Multi-user Login, Emergency Services support, USB device support, Wide-band Codec, Provisioning and Licensing.

The SIP software Release 3.0 for IP Deskphones also continues to improve the overall quality of the IP Deskphone software through the delivery of ongoing resolution of CRs. Numerous quality improvements have been delivered and 9 customer cases have been closed in SIP 3.0.

I’ve only performed very limited SIP testing with the 1120E, 1140E, and 1220 IP phones in non-production environments. I did notice a few feature called “Multi-user Login” which allows a SIP IP phone to connect to multiple SIP servers at the same time. Here’s the blurb from Avaya on the feature (it’s a direct quote from the release notes);

Multi-user Login

The Multiuser feature in SIP Release 3.0 allows multiple SIP user accounts to be in use on the IP Deskphone at the same time. Multiple users, each with their own account, can share a single IP Deskphone allowing each user to receive calls without logging off other users. One user can have multiple user accounts (for example, a work account and a personal account) active at the same time on the same IP Deskphone. You can register each account to a different server, and for each account, the IP Deskphone exposes the functionality available to that account. One account is considered a primary account and is used by default for most IP Deskphone operations. Each account is associated to a line key; the primary account is always on the bottom right line key of the IP Deskphone (this is the first key, Key 01), and an arbitrary key (including a key on an Expansion Module) can be selected for additional accounts.

The following operations are supported:

• Start dialing
• Place a call using the corresponding user account
• Answer an incoming call targeted to that account
• Initiate a call without pressing a line key (for example, by dialing digits at the idle screen and lifting the handset) uses the primary account.

A running IP Deskphone is associated to a single profile that represents one configuration of the IP Deskphone with all relevant persistent data such as preferences and call logs. A different profile is associated to each account used as a primary account. The IP Deskphone can store up to five different profiles; the IP Deskphone takes data from the profile associated to the current primary account. A number of configurations are independent of profiles and tied directly to an account making them available to that account regardless of the primary account you use (for example, voice mail ID).
The IP Deskphone receives and answers calls targeted at any of the registered accounts; the incoming call screen indicates who the call is for. You can place an outgoing call using any of the accounts; the account that you use is displayed on the dialing screen. When a call is active, information from both local and remote parties appear on the screen.
Regardless of which account receives the call, incoming call logs, outgoing call logs, and instant messages appear in a single list. The IP Deskphone indicates the local user in the detailed view of the entry.

Some features are only available to the primary account, such as instant messaging, retrieving parked calls by token, and establishing ad-hoc conference calls.
Please refer to the product bulletin and the release notes for all the details.

Cheers!

After digging through the net and all my documentation I came up with the following reference in one of the UNIStim release notes.

A constant humming noise is sometime heard through the headset when either the Enhanced USB Headset Adapter or the Mobile USB Headset Adapter is connected to the 1120E, 1140E, 1150E or 1165E IP Deskphone. The humming noise heard within the headset can be corrected by upgrading the Headset Adapter firmware to version 2.00.98 or greater.
The USB Headset Adapter firmware version 2.00.98 is available for download from the “Software Download” link under “Support and Training” on the Nortel website located at: http://support.nortel.com. The firmware is available for the 1120E, 1140E, 1150E and 1165E IP Deskphone models under “Phones, Clients and Accessories” as file Adapter3v2.0098.zip.

To load the version 2.00.98 firmware onto the USB Headset Adapter perform the following procedure:

1. Download the firmware file Adapter3v2.0098.zip from the Nortel Technical Support web site
2. Load the file Adapter3v2.0098.zip onto a PC
3. Uncompress (unzip) the file to obtain Adapter3v2.0098.exe.
4. Connect the USB Headset Adapter to the PC
5. Start the Adapter3v2.0098.exe application to load the firmware onto the device.

Hopefully this helps someone else out. It took me quite sometime to locate any reference and I had almost given up.

I’ve placed copies of the zip archive and the readme file on my server.

Cheers!

• 0621C7G for 2007 IP Deskphone
• 0623C7M for 1110, 0624C7M for 1120E, 0625C7M for 1140E, 0627C7M for 1150E and 0626C7M for 1165E IP Deskphones
• 0627C7M for 1200 Series IP Deskphones
• VPN Configuration Wizard release 01.00_00.25

The major change in this software release is the re-branding of the IP deskphone to Avaya from Nortel.

UNIStim 4.2 is the minimum software release that includes changes related to re-branding of the IP Deskphone software to Avaya from Nortel. All instances of Nortel branding within the IP Deskphone software including the start-up splash screen, User Interface elements, and Certificates have been changed to Avaya branding. In addition, the VPN Configuration Wizard software has been rebranded to Avaya

Please refer to the release notes and the product bulletin for complete details.

Cheers!

Update: Monday August 16, 2010

Avaya has re-released the bulletin because of a typo in the document.

You can enable the SSH console by the following commands;

1. Press the Services key twice in quick succession
2. Select Local Diagnostics
3. Select Advanced Diag Tools
4. Place a checkmark in the box labeled Enable SSH
5. Set the UserID
6. Set the Password
7. Apply the settings

There is no reboot required to enable the SSH console. Here’s a quick example of the help command;

[root@centos ~]# ssh 10.1.1.10 -l admin
The authenticity of host '10.1.1.10 (10.1.1.10)' can't be established.
RSA key fingerprint is 09:14:95:11:c2:e6:d7:93:98:2c:4e:ce:e4:2c:64:cc.
Are you sure you want to continue connecting (yes/no)? yes
Warning: Permanently added '10.1.1.10' (RSA) to the list of known hosts.
admin@10.1.1.10's password:

Welcome to Nortel problem determination tool.

You are connected to IP Phone 1120E.
HW version:18001365FF5E4FFFFF
FW version 0625C7F
MAC 001365FFFFF
IP 10.1.1.10

Type "pdtHelp" for list of available commands.
Bluetooth address 00140D01635B

Type "pdtHelp" for list of available commands.
Type "bye" to exit current shell.

PDT> pdtHelp

pdtHelp                                      Print PDT shell help
setLogLevel           <loglevel>             Set LogLevel, Critical:1, Major:2, Minor:3, Warning:4, Info:5
setRecoveryLevel      <recovery level>       Set RecoveryLevel, Critical:1, Major:2, Minor:3
setAutoRecoveryFlag   <flag>                 Set auto recovery flag, turn on:1, turn off:0
printLogLevel                                Print current logLevel, Critical:1, Major:2, Minor:3, Warning: 4, Info:5
printRecoveryLevel                           Print current recoveryLevel, Critical:1, Major:2, Minor:3
printAutoRecoveryFlag                        Print auto recovery flag
printUptime                                  Print set uptime
printLogFile          [severity level]       Print log files; Args - Critical:1, Major:2, Minor:3, Warning:4, Info:5
clearLogFile                                 Clear content of error log file
taskMonShow                                  Show task monitor list
taskMonAddTask        <taskName | task id>   Add a task to task minitor
taskMonRemoveTask     <taskName | task id>   Remove a task from task monitor
setCpuSamplingPeriod  <value>                Set CPU sampling period, range: 180-360s, step 10s
i                                            Print all task Info
ti                    <taskName | task id>   Complete info on TCB for task
tt                    <taskName | task id>   Task Trace
memShow               [level]                Show system memory partition blocks and statistics
checkStack            <taskName | task id>   Print a task's stack usage
ls                    [dirname] [-f]         List contents of directory, -f: include details
lsr                   [dirname]              Recursive list of directory contents
cd                    [dirname]              Set current working path
usbFsShow                                    Display MSDOS volume configuration data of USB memory stick
usbls                 [dirname] [-f]         List contents of USB directory, -f: include details
usblsr                [dirname]              Recursive list of USB directory contents
usbcd                 [dirname]              Set current USB working path
pwd                                          print the current default directory
ping                  <host ip> [# of pings] Test that a remote host is reachable
tracert               <host ip> [max hops]   traceroute to any host
netinfo                                      Print common network info
routeshow                                    Display host and network routing tables and stats
arpShow                                      Display entries in the system ARP table
listcerts                                    List all trusted certificates
printcert             <index>                Print a trusted certificate in detail
listcrls                                     Prints a detailed list of CRLs
listdevcerts                                 Prints all device certificates
listsecuritylogs                             Lists all events logged through the security interface
securitypolicy                               Prints the current Security Policy values
gxasinfo                                     Lists the GXAS configuration and current status
reportWidgetData                             shows widgets info
reportWindowData                             shows windows hierarchy
turnOnScreenScrape                           turn Screen Scrape feature on
turnOffScreenScrape                          turn Screen Scrape feature off
setScreenScrapeDelay  <delay>                set delay in ms for the Screen Scrape process
sendKey               <code> <state>         Emulate key with a code "code". State 0/1/2 = Key Down Message/Key Up Message/Key combination down followed by up.
showVPNStatistics                            Show VPN Statistics
showVPNStatus                                Show VPN Status
showVPNFilter                                Show VPN Filter
setVPNLogLevel        <loglevel>             Set VPN Log Level - 0:turn off log/1:log info/2:log info,error/3:log error,debug,info
printVPNLogLevel                             Print current VPN logLevel - 0:turn off log/1:log info/2:log info,error/3:log error,debug,info
showFIPSStatus                               Show FIPS Status
setVPNNatKeepaliveIntervalOverride <interval>             Set VPN NAT Keepalive Interval Override
scrShow                                      Show SCR Status
printSetInfo                                 Print HardwareID, FirmwareID and MAC address
vxshell                                      Switch to vxShell
bye                                          Exit current shell
PDT>

I don’t know that I would advise someone to enable this feature on every IP phone they deploy but it can certainly be helpful if enabled when needed during troubleshooting.

Cheers!

The following enhancements are now available;

The following issues have been resolved;

• (091210-67218/091216-70624) 2050 is not marking QoS values in packets (see Technical Advisory Section)
• (091030-43866) IP Call Recording stops working after retrieve call from Hold
• (091203-63291) Unable to configure annotation feature keys in Expansion Module.
• (091008-30561) i2050.exe process does not release GDI Objects
• (100224-05132) 2050 IP Softphone on CICM: Delay when hanging up from a call
• (100310-13208) Configuration tool does not support customized skins
• (100312-14320) Configuration tool errors when applying Node/TN info. The configuration tool will now allow empty values for any parameter in the config.ini file and will not overwrite the registry entry if it is blank. This allows customers to upgrade to new configuration settings without changing selected values such as the Node/TN information that may already be set in the registry.
• (100405-25547) Inconsistent behavior of IP 2050 Settings for first launch if ConfigurationTool
• (090812-96650) GN Netcom 8110 USB adaptor does not work on the docking station

Thankfully Avaya/Nortel has already provided plenty of configuration examples for a number of well-known carriers.

2010-00000227_1.1_BCM50_BCM450_R5_Configuration_Verizon.pdf
NN10000-103_Ver_1.4_Nortel_BCM50_3.0_SIP_Config_Guide.pdf
2009-00002459_1.0_BCM_5.0_Configuration_Guide_Skype_SIP.pdf
2010-00000229_1.0_M50R3_M450R1_Configuration_Guide_For_Bell.pdf
2010-00000219_1.0_Config_Guide_Bell_SIP_Trunking.pdf
2009-00002460_1.1_BCM ConfigurationGuide_PAETEC_SIP_Trunking.pdf

Have a look at the above documents if you are searching for how to configure the BCM50/BCM400 for SIP trunking to a public provider/carrier for PSTN access.

Cheers!

• 0621C7D for 2007 IP Deskphone,
• 0623C7J, 0624C7J, 0625C7J, 0627C7J and 0626C7J for 1110, 1120E, 1140E, 1150E and 1165E IP Deskphones respectively and
• 062AC7J for 1200 Series IP Deskphones

UNIStim 4.1 adds the following features;

• UNIStim 4.0 functionality delivered onto the 1165E IP Deskphone
• Quality improvements to Secure Signaling using DTLS
• Adjustable open-microphone warning tone during Zone Paging
• UNIStim VPN client interoperability extended to include Avaya VPN Gateways

The two product bulletins covering the software release can be found here and here.

Notice any formatting differences between the two bulletins?

Cheers!

I should warn folks that the phone is sold with different SKUs depending if you want it running the UNIStim or SIP protocol. Upgrading the phone between the UNIStim and SIP firmwares is not supported by Avaya/Nortel. With that said I was successful in upgrading/converting a UNIStim SKU’d phone with the SIP firmware available from Avaya/Nortel’s Software Communication System (SCS). I did have some issues downgrading/converting the same set back to UNIStim, although I eventually found the workaround that was needed to trick the SIP firmware into believing I had newer firmware. I can share that with anyone that is interested or if anyone is stuck in a similar position.

The default configuration password is:

26567*738

Cheers!

Update: Monday February 22, 2010

It might be easier to remember the password as follows:

COLOR*SET

I should point out that this document covers the Nortel Succession Call Server 1000 v6.0 software. While this document obviously can’t cover every possibility it does a great job of getting your feet wet and is welcome addition to my library.

You can find the document here.

Cheers!