VoiceOverIP

Asterisk Now with Avaya IP Phones

0
January 15, 2012
by Michael McNamara in ,

There’s been a lot of discussion lately around connecting Avaya (legacy Nortel) IP phones with third-party SIP capable call servers. I’ve personally toyed with Asterisk on a number of occasions and have always been impressed so I recently setup an Asterisk Now installation (AsteriskNOW-1.7.1-i386.iso) on a CentOS 6.2 KVM host so I could re-test the interoperability between the latest version of Asterisk (v1.6.2.20) and the 1100 and 1200 series IP phones from Avaya running SIP v4.0 and SIP v4.3 respectively.

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!

SIP Software Release 4.3 for 1100/1200 Series IP Phones

6
December 14, 2011
by Michael McNamara in ,

Avaya has released SIP software release 4.3 for their 1100 and 1200 series IP deskphones.

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!

UNIStim Firmware Release 5.3 for IP Deskphones

2
December 6, 2011
by Michael McNamara in ,

Avaya has released UNIStim firmware 5.3 for their IP deskphones;

  • 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!

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