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!
Hello Mike,
Great info, thanks. But what if you want to enable the extended features, like multiple logins, or the bluetooth function on a 1120 or 1140 with Asterisk. What kind of license do you need for something like that?
Thanks
When can I get the SIP firmware for the Avaya IP Phones?
You’ll find the SIP firmware on the Avaya support website.
https://support.avaya.com/downloads/downloads-by-contenttype.action?product_id=P0599&product_name=1100+Series+IP+Deskphones&release_number=SIP+4.x
Good Luck!
Michael thanks so much for the great posts. I want to purchase some of the Avaya 1120e phones as I really like the look of them but there’s 2 things I’m not clear on, hoping you can help.
1) In the above examples, it all seems to deal with just 1 phone. How does it work if I have a TFTP server and I have multiple phones? I’m used to Cisco and Polycom where the config file is the MAC address of the phone, but the above example says the phone will look for the 1120eSIP.cfg file and then users.dat file. I’m not clear on how 5 phones connecting each get their own config info?
2) What are the limitations of SIP? I keep seeing no advanced features without an Avaya licence, but what are those advanced features? Like can I still conference call, transfer, blind transfer, etc or is it REALLY limited as in can only make/receive calls etc?
Thanks so much!
Hi Lee,
You are very welcome…
1) You can create a configuration file using the MAC address of the IP phone similar to Cisco and Polycom. In the example above the user or administrator is configuring the SIP TN and password. The IP phone will store this information and will not ask for it over subsequent reboots. You can use auto-provisioning for everything if you desire, just need to setup the TFTP files appropriately.
2) There’s a discussion somewhere regarding the Advanced License token… Bluetooth, Encryption, VPN, Multiple Logins, etc.
You’ll find what your looking for in this post;
http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip/
Good Luck!
Hi,
I connect successfully to Asterisk. When on the phone (both, incoming or outgoing) the phone will disconnect every 10 secs and display “logging on user..” sip trace on Asterisk doesn’t show much. Please help. Thx
Hi Len,
I’ve never seen that problem myself. I would suggest you start by looking at the logs on the Asterisk server. If you run a constant ping between the devices you’re not seen any drops right?
Sorry I can’t help more.
Thanks for responding. Asterisk logs just shows that it is timing out. Constant pinging the phone doesn’t drop any packets. SIP_PING YES doesn’t help either.
The phone is behind NAT but audio works nicely before call drops.
I would guess your issue is going to be NAT… you probably need some NAT keepalive or something like that. If you look at your router your NAT session is probably being aged out of the NAT table which causes the connection between the Call Server (Asterisk) and IP phone to break and hence the reboot.
Cheers!
Thanks again. Yes it is indeed an issue with NAT. I connected successfully w/o NAT and didn’t have the disconnect problem.
I’m thinking its rather the phone then the router as all other IP Phones (Cisco, Aastra, Polycom, Grandstream etc,) don’t have this issue.
I believe the way most IP Phones “keep alive” is by sip “options” messages (unless one specifies a STUN server). Perhaps Avaya sip f/w is not pure sip. I will run a trace and see.
Best regards
Hello sir,
Useful articles, thanks!
We are trying to connect a 1220, it says looking for DHCP, then attempting TFTP, then connecting to S1
When you say “I did have to re-configured the 1220 to AllAut before it would honor the settings in the TFTP provisioning file.” What do you mean? Is this a setting on the phone, or in the config?
Thanks
Dave
Hi Dave,
While configuring the 1100 and 1200 series IP phones you’ll have an option labeled “AllAut” (All Automatic) to select on the IP phone itself. This will configure the IP phone so it utilizes all automatic options.
Good Luck!
hi,
we have avaya 5621 and 5620 IP phones and I’m trying to connect them to AsteriskNow 2.0.2. Their setup is different than what was mentioned in the post as the phone is looking for some i20d01a2184e.bin and other .src files. how to obtain these and where to place them in the asterisk server. Please tell me what values to give for port,fileserver and router when press the * to program in the phone. I’ve tried all the combinations i know but it doesn’t seem to work. Following are the /etc/asterisk/sip_custom.conf and /etc/asterisk/extensions_custom.conf file entries.
In phone, at some point shows error HTTP – 1 404, and stops at the screen Discover 192.168.0.186 (which is the asterisk server IP)
Please help!
/etc/asterisk/sip_custom.conf
[phone1]
type=friend
host=dynamic
port=1720
secret=200
context=users
deny=0.0.0.0/0
permit=192.168.0.167/255.255.255.0
[phone2]
type=friend
host=dynamic
secret=202
context=users
deny=0.0.0.0/0
permit=192.168.0.0/255.255.255.0
/etc/asterisk/extensions_custom.conf
[users]
exten=>200,1,Dial(SIP/phone1,20)
exten=>202,1,Dial(SIP/phone2,20)
Hi Eranga,
Unfortunately I don’t have any Avaya 5600 series IP phones to test and I’ve actually seen them myself.
If Avaya provides one or two I’d be happy to document the process but short of that I’m afraid I can’t really be of any help.
Sorry.
Hi Michael,
Thanks for your response. If you can get your hands on Avaya 5620 phones, please do let me know how to go about it as i’m really stuck not knowing how to proceed. I hope the Avaya would provide you the phones and at least let you help us, because there’s no proper support or documentation by Avaya at all on these. I’m so disappointed about their service. Keep up the good work!
cheers,
Eranga.
Hello
By residential Belkin router doesn’t allow for DHCP option 66. Is there a way I can upload the SIP firmware such as through the local USB port on the 1120e?
Thanks in advance for your response.
Michael
By chance do you know how to setup a 4621sw or a 9650 to use elastix ? I’ve already got it to upgrade to sip but discovery is to my public address not my server addresss.?
Hi Robert,
Sorry I wouldn’t really know… I would expect you need to configure the IP address and port of your SIP server into the phone.
Good Luck!
Hi
I’m trying to connect my Avaya 9650 (SIP, Firmware 2.6.10) to Asterisk 1.8.13.1-0006, but it doesn’t work. My Asterisk ist connected with the Voip-Provider, but the Phone can’t find de User (6000). By login, the phone display’s “Acquiring Service” and Asterisk says, that my phone ist unavailable. I have submitted in the extensions.conf the MAC-Adress of my Phone and the IP, Protocoll is TCP.
Can somebody helps me? (in English or German) ;-) thanks …
You may want to double check if TCP enabled and binded in sip.conf ; and in Avaya 9650 , set SIP Proxy to your asterisk .
do you know of anyone getting the 9611g to work with asterisk? I have updated my phones with the latest SIP firmware, and I can get a dial tone on the phone but I can not call any extensions and when I dial a landline/cell number the phones do call and connect but niether party can hear anything.
also there is a error icon in the upper left corner of the phone screen.
Unfortunately I haven’t played with the Avaya 9600 series IP phones much so I can’t really say personally.
Good Luck!
Hi Michael,
I’m facing a big Issue with my Avaya Phones Type 4621 connected to Elastix Server (Asterisk 2.8). Problem: when I register one of the Phones, they loose Registration after approximately 12 hours. Then I have reboot the phone everytime. I there a workround for this Issue, such as a automaique Reregistraio`n? Thanks a lot for Your Help.
Regards Andy.
Hi Andy,
I haven’t been doing a lot with Asterisk and/or SIP handsets lately… I’m guessing that there’s an issue with the SIP registration renewal or refresh timer. Sorry I can’t be of more help.
Good Luck!
I know this is an ancient blog post, but do BLF (busy lamp field) indicators work on the Avaya phones when using asterisk?
Unfortunately I don’t know… you are referring to multi-line appearances on the IP phone? I doubt that SIP supports that so it would likely only be available via the H323 protocol.
Cheers!
Hello, please can you help me with a 1608 phone to connect them to SIP, I have 2 questions:
1. One of the phones remains loaded the Firmware and it is restarted continuously and it will not be removed from that state, is there a sequence of numbers when turning on the phone to reset it to the factory ?.
2. In which part of the configuration file the Mac of the equipment is specified to generate independent files with its extension and that each telephone takes it independently.
I appreciate the answer Michael.
Hi Ivan,
I’ve only ever done that with the legacy Nortel (Avaya Blue) IP phones, 1100 series specifically.
The IP phone would request via TFTP a configuration filename from the server using the IP phones MAC address as the filename.
I’ve never had the opportunity to test this feature using the Avaya Red IP phones, 4600, 9600, 1600 series IP phones.
Cheers!
Hi i have avaya 9608 an 9611G i can connecto to asterisk with elastix my problem is this when i call from one avaya to another phone i only see the number of the extension not the name of the extension i wan to see both name and number can you tell me if there is and option to enable to see the name of the extension in both models
Hi Mauricio,
Unfortunately it’s been a long time since I played around with Asterisk and the UNIStim module… I don’t believe I could be much help today.
Sorry
Hi Michael thanks for your answer your post give a few Ideas thanks i will try to solve this