Posts tagged 1220
Asterisk Now with Avaya IP Phones
0There’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 3.2 for IP Deskphones
11
Avaya has released SIP software release 3.2 for their 1100 and 1200 series IP deskphones. This release adds support for the 1120e, 1140e, 1165e, 1220, and 1230 model IP deskphones.
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!
Nortel IP Phone 1200 Series
45
We recently purchased two Avaya/Nortel 1220 IP phones for testing in our environment as a possible replacement to the manufacture discontinued i2002/i2004 IP phones. We’re evaluating whether we should purchase the 1120e/1140e or the 1220/1230 as our standard IP phone going forward. An obvious concern going forward is that the phone support the Session Initiation Protocol (SIP) so that it will be potentially capable of inter-operating with whatever soft switch or PBX we might have in the backend, be it the Avaya Aura or the legacy Avaya/Nortel Call Server 1000.
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:

