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!

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!

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!

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!

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!

• Communication Server 1000 6.0 SIP Trunking with Quest
• Communication Server 1000 6.0 SIP Trunking with Colt
• Communication Server 1000 5.5 SIP Trunking with AT&T VoEVPN
• Communication Server 1000 6.0 SIP Trunking with Verizon Business

I’ve been very busy as of late with work and hope to write some original posts very soon.

Cheers!

I figured I’d better not skip the Bell Canada document for fear of upsetting my Canadian readers. ;)

I’m really excited to see Avaya providing this information to both their resellers and customers. I’d like to thank those that wrote and lobbied for these documents within Avaya. It’s extremely encouraging to see Avaya willing to empower their users so they can leverage their products and investments to their fullest.

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!

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!

I ordered the service online and it took about a 11 days for the Linksys PAP2 to arrive. I had to call four days after I placed the order to find out the status since there was no email message informing me that the equipment was back ordered).

The installation of the Linksys PAP2 was quite easy. I just connected it to the Verizon Actiontec router and plugged the RJ11 jack into my phone. Within seconds I had dial tone from the Linksys PAP2. I didn’t need to make any changes to the Verizon Actiontec router although it might be necessary later to apply some QoS settings.

It took 7 days to port my original AT&T CallVantage phone number to Voice Pulse. Prior to porting my phone number I just setup CallVantage to forward all calls to the temporary number assigned by Voice Pulse.

There are an amazing number of call routing and call filtering features including telemarketer block which promises to block automated and computerized dialing services used by a vast number of telemarketing companies.

So far the service has been great and very reliable. And would you know that I actually received a phone call from Voice Pulse confirming the port of my home phone number. And I even spoke to an actual human being that I could clearly understand. Did I mention that they called me?

If you have reliable Internet broadband and your looking for a good Internet phone provider you won’t go wrong with Voice Pulse. Voice Pulse will also provide you with SIP trunks for your Asterisk deployment.

Cheers!