Posts tagged SIP

Asterisk Now with Avaya IP Phones

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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

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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!

SIP Software Release 4.1 Service Pack for IP Deskphones

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Avaya has released SIP software release 4.1 Service Pack (4.01.15) for their 1100 and 1200 series IP deskphones. I’ll refer you to the release notes for the 4.1 Service Pack software release for the all the details. Let me also reference the 4.1 (4.01.13) software release notes since I never announced it here.

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!

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