Over the past weekend I set out to setup Asterisk, an open source communication server, to test some of the issues reported in a thread over on the discussion forums. I had an Avaya 1120e and 1165e IP phone available to test with, however, both phones were running the UNIStim software for the Avaya Communications Server 1000. I needed to upgrade them to support SIP.
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.
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;
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.
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.
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.