Disabling features from extended feature set on IP Deskphone

[hexanol]

Greetings to all,

I have an Avaya 1220 IP Deskphone (also known as Nortel 1220 IP Phone) and I would like to use it on a non Avaya call server. The phone is using the SIP firmware in version 3.2, and I've also done some tests on the new 4.0 SIP firmware.

The problem is, since it's not connected to an Avaya call server, all the features from the extended feature set must be disabled or else the phone is not usable. And by default, there's a good amount of these features enabled.

Now, I'm configuring the phone via the configuration files, but I've not been able to deactivate the 'Authentication Security' and 'Signaling Security (TLS)' extended features.

I do have been able to disable these features via the phone user-interface, by going in the 'device settings' menu and setting 'SIP TLS Port' to 0. But I've not been able to do so via the configuration file, even by using a 'SIP_TLS_PORT 0' line, the phone just seem to ignore it.
 I've also tried setting 0 to parameters SERVER_TLS_PORTX_Y and SERVER_TCP_PORTX_Y (where X and Y are numbers), with no result.

What's worst is that I've tried with the new 4.0 SIP firmware, and now, it looks like there's not anymore the 'SIP TLS Port' option accessible via the phone user interface (!).

I would really like to be able to turn off these features from the configuration files and not have to play on the phone menu to disable them (especially since it seems like it's not possible anymore to do so with the new firmware). Could someone help me ?
 I would like to ask my question to Avaya but it seems to be a bit hard if you don't have a support contract with them (they have a forum, but you can't post on it).

Thank you

[Michael McNamara]

I was going to point you to this thread but it was you that posted the solution so I'm guessing that won't really help much.

I will ask a few Avaya/Nortel folks to read and comment (either publicly or to me in private which I will then reply).

Cheers!

[hexanol]

I was going to point you to this thread but it was you that posted the solution so I'm guessing that won't really help much.

Indeed, I know this thread pretty well.

The thing is that the situation is worse with the new 4.0 SIP firmware, where you can't even disable all the extended features from the phone UI. I'm quite sure there's a way (maybe not officially documented) to disable all the extended feature from the configuration files, but I've done a good amount of tests on it and that said I'm still looking for the answer (as you can see) and this forum is actually what I found best to get a bit of help/support.

Thanks for your help

[Michael McNamara]

I've spoken to an Avaya resource and they're looking into it...

Cheers!

[hexanol]

Sweet! You have better contact than I do as I can see...

[Michael McNamara]

Let's not get to excited until there's an answer.

I have a 1165e IP phone here that I can try and setup with a virtual Asterisk install... I'm guessing the 1200 and 1100 series probably behavior about the same with respect to the SIP 4.0 release and the new licensing.

That might be fun to try.

[Michael McNamara]

I had a little fun this morning and stood up an AsteriskNOW installation on my VMware ESXi host. I took an 1165e and an 1120e and converted them from UNIStim to SIP and was able to craft some basic configuration files to get them to connect to the Asterisk server.

I haven't really utilized the 1165e (Thanks Gord!) up until now so it looks like I'm running on an evaluation license. I had previously utilized the 1120e to test the UNIStim VPN Client (UVC) connecting it to my home/office network and configuring it to connect to my work CS 1000 via a Nortel VPN Router 1700. When the 1120e attempted to connect to the Asterisk server it displayed the following;

Information
Could not obtain license
Login has been denied


I used a very basic 1120eSIP.cfg file to start with;

[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]


With the following Users.dat file;


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

ADMIN_PASSWORD 26567*738


So let me embark on trying to fight the right combination of settings in the configuration files.

Cheers!

[Michael McNamara]

That didn't take too long at all.... so I modified my users.dat file to include the following;

# 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 also had to go to the IP phone and set all the configuration settings to Automatic;

Press the Services key twice in quick succession, select Device Settings, enter the admin password (default is COLOR*SET) select the Auto, select AllAut, select Config, select Apply and the IP phone will display "Please wait..." and start to reboot itself.

When the IP phones rebooted they both connected to the Asterisk system and were online.  When I checked the licensing information I found the following on the 1165e;

License Tokens
   License mode: Node locked
   License status: No request
   Token type: Evaluation
   Tokens requested: 0
   Tokens acquired: 0
   Token expiry: 31 days

And on the 1120e I found the following;

License Tokens
   License mode: Node locked
   License status: No request
   Token type: Evaluation
   Tokens requested: 0
   Tokens acquired: 0

I realize you specifically mentioned a 1220 IP phone but I'm guessing the 1100 series and 1200 series are pretty similar and I believe they run the same software (perhaps someone from Avaya could fill us in). I do have access to a 1220 IP phone but it's not in front of me at this time.

Cheers!

[hexanol]

Press the Services key twice in quick succession, select Device Settings, enter the admin password (default is COLOR*SET) select the Auto, select AllAut, select Config, select Apply and the IP phone will display "Please wait..." and start to reboot itself.

Great! That part did it for me.

In fact, what's weird is that every settings were already set to 'automatic', i.e. I didn't had the 'AllAut' softkey but only the corresponding 'AllMan' softkey. So I went back to the 'device settings', menu, pressed Apply (although nothing had changed), the phone rebooted and it then took into consideration the 'SIP_TLS_PORT' settings in my configuration file it previously didn't want to. Now, even if I reset my phone, I don't have to do this manipulation again.

Whoa. So I guess we can mark this thread as solved! Thank you.

[Michael McNamara]

I got my hands on a 1220 IP phone and sure enough after configuring it just like the 1100 series IP phones it's using the evaluation license with the following to advanced features enabled;

• Authentication Security
• Security Security (TLS) enabled

The 1220 IP phone had been running the SIP 1.0 software release until I just upgraded it tonight.

Let me see if I re-apply the AllAut setting in the phone configuration.

(10 minutes later)

That seems to have done the trick... the License Status now reads No request, previously it read Active. I will note that it took an extermely long time for the 1220 to connect to the Asterisk system. I would probably guess somewhere around 3-4 minutes, it just sat there saying "Logging in..."

So it wasn't you hexanol it was technology run a muck...

Cheers!

[rlsmeridian]

Michael,

Are the files and code in posts 6 and 7 above in the firmware downloads from Avaya's site?

I tried modifying the configuration many times in the past and hopefully this does the trick.....I must be doing something incorrect. 

So if I follow the basic instructions above in this discussion, I should be able to get the phone working?

If I can get it working, I am going to buy myself an 1165E.  Anywhere you'd recommend for good pricing?

Thank you,

Joe

[Michael McNamara]

Hi Joe,

Well buying an 1165e won't get you any further... it runs the same SIP 4.0 software as the other 1100 series IP phones and has the same licensing model. You can certainly get any 1100 series IP phone to behave as a simple SIP handset. If you want some of the extended features then you need to purchase a license.

I have an 1120e, 1165e and 2 1220 IP phones sitting next to me connected to an Asterisk IP PBX via SIP and they work fine... no licensing issues.

Just follow the basics above... you may have to hard set the TFTP server via the handset as oppose to using DHCP.

Cheers!

[rlsmeridian]

Hi Michael,

I like the color display on the 1165E.  I have 3 Unistim 1140E's running on a BCM50 RLS6.  I also have 1 1140E and 1 1120E SIP phone running directly off of VOIP.MS, my SIP trunking provider. 

When configuring the 1140/1120 phones, I went in and updated the configuration file, but I didn't upgrade the .dat files that you are explaining above, so that could be where the problem is.  I actually don't want to use the extended feature set, just the basics will be fine. 

The Nortel sets are so easy to use and provision, well except this feature we are trying to figure out above.  I also have a Cisco 7970, well, indirectly have it.  I bricked it last week by uploading some code that was corrupt.  I sent it back to the reseller and they will hopefully give the phone CPR and send it back.  This time I will have to be more careful loading the TFTP server and not crash the phone!  I will admit that the Nortel/Avaya SIP sets are much easier to provision than the Cisco sets!  After my debacle this weekend with the complicated Cisco code, the notes above should be easy.   I am also running an Avaya SIP 9630 set on VOIP.MS.  When I get the 7970 set back Friday, I hope to successfully get VOIP.MS working off it. 

So your remarks in blue above, are all I have to send to the phone via TFTP server?

Thank you,

Joe

[Michael McNamara]

Yes.... you just need two files really... SIP1120e.cfg and users.dat (or whatever you want to call it).

The IP phone will automatically try to download the SIP1120e.cfg (substitute the model of IP phone here - example, SIP1165e.cfg, SIP1140e.cfg, SIP1220.cfg, etc). The IP phone will parse that file and then load users.dat or whatever you call the file. You should make sure that all the configuration options on the IP phone are set to AllAut.

Here's a small reseller that I've used before... very good;
http://www.robertscommunications.net/customer/home.php

Good Luck!

[rlsmeridian]

Hi Michael,

I tried everything above, but still no luck, I will copy and paste my Files and please let me know if you see that I am doing something incorrect?

The phone seems to not be reading my Device Config. file.  I was successful in loading the 4.0 software, but back to 2.2 until I get this figure out.

Thank you,

Joe

DeviceConfig.dat
[FW]
DOWNLOAD_MODE AUTO
VERSION SIP1140e04.00.04.00
FILENAME SIP1140e04.00.04.00.bin
PROTOCOL TFTP

[DEVICE_CONFIG]
DOWNLOAD_MODE FORCED
VERSION 000200
FILENAME Users.dat

[DIALING_PLAN]

DNS_DOMAIN joesus.net
SIP_DOMAIN1 voip.ms
SERVER_IP1_1 192.168.1.2
SERVER_IP1_2 64.120.22.242
SERVER_PORT1_1 5060
SERVER_PORT1_2 5060
SERVER_RETRIES1 3
SERVER_RETRIES2 3
DEF_USER1 116477_1140

VMAIL 5000
VMAIL_DELAY 300

DEF_LANG English
DEF_AUDIO_QUALITY High

ADMIN_PASSWORD 26567*738

NOW Users.dat file:

DNS_DOMAIN joesus.net
SIP_DOMAIN1 voip.ms
SERVER_IP1_1 192.168.1.2
SERVER_IP1_2 64.120.22.242
SERVER_PORT1_1 5060
SERVER_PORT1_2 5060
SERVER_RETRIES1 3
SERVER_RETRIES2 3
DEF_USER1 116477_1140

VMAIL 5000
VMAIL_DELAY 300

DEF_LANG English
DEF_AUDIO_QUALITY High

ADMIN_PASSWORD 26567*738

# 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