Here's what I did to get them to work.
This is a document in progress. I"m just typing down my settings and findings as I go through it, and will clean it up.
First 9608 phone.
Testing FreePBX Distro 13
Took a phone reset the configs to factory
I put 46xxxsettings.txt, 96xxupgrade.txt and SIP97xx_2_6_9_1.bin into var/www/html in my freepbx server
Powered on the phone
Factory login/password
* = enter config file (sometimes it a button marked "PROGRAM" on the screen)
craft = password
I hard coded an IP/mask/gateway (router)
HTTP/S server i put in as my asterisk server (sharing the html folder for its configs)
Everything else was auto
SIGNALLING = SIP
SIP GLOBAL settings:
SIP DOMAIN=IP of pbx
Avaya Environment = NO
Reg Policy =simultaneous
Failback =auto
CFG SERVER= IP Of PBX
USER ID FIELD =no
SIP PROXY = IP of PBX
Transport=tcp
sip port 5060
In asterisk i created an extension 1000 with a password of 1000 (unsecured for testing, server not net accessible anyway)
On the FreePBX configuration is configured the SIP settings to ensure that my network was one that would be recognized (my phone for testing are on a different vlan) and in SIP settings in the FreePBX gui i set TCP = YES
Save the changes and boot up the phone.
The phone should register and show login
Type in my login and password, phone registered.
-----
TODO:
working on how to self configure the phone. If i type in the user/logon manually, then reboot the phone, it will autologin after, but i want the user/secrete to be deployable from a HTTP server or something.
Some Firmware
This is a document in progress. I"m just typing down my settings and findings as I go through it, and will clean it up.
First 9608 phone.
Testing FreePBX Distro 13
Took a phone reset the configs to factory
I put 46xxxsettings.txt, 96xxupgrade.txt and SIP97xx_2_6_9_1.bin into var/www/html in my freepbx server
Powered on the phone
Factory login/password
* = enter config file (sometimes it a button marked "PROGRAM" on the screen)
craft = password
I hard coded an IP/mask/gateway (router)
HTTP/S server i put in as my asterisk server (sharing the html folder for its configs)
Everything else was auto
SIGNALLING = SIP
SIP GLOBAL settings:
SIP DOMAIN=IP of pbx
Avaya Environment = NO
Reg Policy =simultaneous
Failback =auto
CFG SERVER= IP Of PBX
USER ID FIELD =no
SIP PROXY = IP of PBX
Transport=tcp
sip port 5060
In asterisk i created an extension 1000 with a password of 1000 (unsecured for testing, server not net accessible anyway)
On the FreePBX configuration is configured the SIP settings to ensure that my network was one that would be recognized (my phone for testing are on a different vlan) and in SIP settings in the FreePBX gui i set TCP = YES
Save the changes and boot up the phone.
The phone should register and show login
Type in my login and password, phone registered.
-----
TODO:
working on how to self configure the phone. If i type in the user/logon manually, then reboot the phone, it will autologin after, but i want the user/secrete to be deployable from a HTTP server or something.
Some Firmware
I recently acquired several Avaya 9650 phone and got them working with the latest SIP Firmware using pretty much the same setup. Was only able to enable basic functions but was curious if there was a way to enable some extended SIP Features that one can normally use with these phones on an Avaya System?
ReplyDeleteI need help on Avaya 1608 to work on isaabel or change to sip I will be grateful if anyone has been successful
ReplyDeleteI have the same demand, as I bought a large amount, but I saw that it doesn't support SIP.
DeleteDid anyone get it?