Avaya 9640G Sip Conversion

We just came into posession of a number of these phones. They are bulky but the layout is nice and clean, they are solid and it would be nice to use them. As with many Avaya sets these are setup for h232 so we need to get these on SIP and programmed. Thankfully Avaya still have the files available so let’s get going…. This will be quick and dirty as its more an aide memoir for each step.

You’ll need a HTTP server at the least and control over the DHCP server. Our Lab is FreePBX and PFSense so this is easy for us.

So first up, reset the phone to defaults. The default password is “CRAFT” but if your phones have another set you may need to do some digging on how to wipe these. I have reset over 100 of these in the last few days and no one had a non default password so the odds are good.

Power up the phone (We used POE) and wait for the DHCP prompt. Hit * and enter the password above (27238). Select clear and let it reboot.  You’ll need to go back into that menu again and scroll down to “SIG”. Change from Default to SIP.

If you are doing just one phone…

Go into ADDR and set the http server to the IP of your web server and then exit and let it reboot (Again), you should now get an error on the phone, “HTTP: 1 -401”. On your web server in the error log you’ll see something like :  [Tue Mar 12 11:54:54 2019] [error] [client 192.168.223.133] File does not exist: /var/www/html/96xxupgrade.txt

If you have a few to do you can use DHCP option 242. Set it as type string and pop in your HTTP server address and some vlan info as follows:

L2Q=1,L2QVLAN=0,VLANTEST=0, HTTPSRVR=<youserverip>,HTTPDIR=/<httpdir>/

Now it DOES seem if doing things this way you CAN specify a directory (see below for why I mention it) and this does work, I’ve verified it. Having made the procedure below work I had a large number of these to do and I was buggered if I was doing them all by hand.

You’ll now need to upload the contents of the firmware from ftp://ftp.avaya.com/incoming/Up1cku9/tsoweb/9600/05152017/96xx-IPT-SIP-R2_6_17-172303.zip

There seems to be no way to tell it where to look on the server doing it manually so sadly, this is going into your web server root unless you use DHCP option 242 as above. It may be possible to to clean this up with virtual hosts of you are so inclined. In my case I took out all the languages I didn’t need.

Either method, you now need to reboot the phone and it should trundle off and update itself. This can take a while and the phone may seem it has died or gotten stuck, be patient. It’ll reboot a few times.  Once it boots the UI is noticeably different, you’ll getr a complaint about no call server and it’ll go into a boot loop. Press the program key when offered to break the loop.

You’ll now need to sort out a settings file. Create the file 46xxsettings.txt in the same location as the other files you uploaded. Pop the contents below in this file BUT make sure you edit things to reflect your setup…

SET DNSSRVR 8.8.8.8
SET DOMAIN <SIP SERVER IP>
SET SIPDOMAIN <SIP SERVER IP>
SET SIPPORT 5160
SET SIP_CONTROLLER_LIST <SIP SERVER IP>:5160;transport=tcp
SET SIPREGPROXYPOLICY alternate
SET CONFIG_SERVER_SECURE_MODE 0
SET SIPPROXYSRVR <SIP SERVER IP>
SET SIPSIGNAL 1
SET SIP_PORT_SECURE 5161
SET ENABLE_AVAYA_ENVIRONMENT 0
SET DIALPLAN [2-8]xxx|91xxxxxxxxxx|9[2-9]xxxxxxxxx
SET PHNNUMOFSA 4
SET SNTPSRVR <NTP SERVER IP>
SET GMTOFFSET -5:00
SET DSTOFFSET 1
SET DSTSTART 2SunMar2L
SET DSTSTOP 1SunNov2L
SET DISPLAY_NAME_NUMBER 1
SET SIG 2
SET HTTPSRVR <HTTP SERVER IP>
SET MSGNUM *97
SET ENABLE_EARLY_MEDIA 1
SET RTP_PORT_LOW 10001
SET RTP_PORT_RANGE 9999
SET SIG_PORT_LOW 5160
SET SIG_PORT_RANGE 1

Note the port, 5160! If you are using CHAN_SIP exclusively or its an older freepbx change this to 5060. If you are on a newer install you’ll need this set to 5160 if PJSIP is your primary channel driver. This is yet another device in the LONG list of things that just don’t play ball with PJSIP. If anyone can make it play please let me know but for now it seems its yet another thing that’s been broken.

Regardless of which port you use, you’ll need to enable TCP for CHAN_SIP. I was able to make this work with UDP, however it was acting up, a little digging shows that this is known to be an issue.

Restart the phone and you *should* get prompted for your username (Extension number) and Password (Secret)

Log in and you should be good.

There is one really handy feature with these, press the menu button and you can logout… this make these phones potentially useful for hotdesking!

Now there are a few other things you can mess with , the settings file is dealt with in depth in a few locations, https://www.3cx.com/community/threads/avaya-96xx-9620-phones.11168/ does have a pile of info on these. There are some known limitations and you can make things play ball a little better if you don’t mind recompiling your freepbx instance, this is covered here: https://community.freepbx.org/t/avaya-96×1-extended-features/40543

DHCP Options are covered here : https://downloads.avaya.com/css/P8/documents/003876932

 

 

 

 

 

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