Hi Michael thanks for your answer your post give a few Ideas thanks i will try to solve this

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 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!

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.

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!

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!

Unfortunately I haven’t played with the Avaya 9600 series IP phones much so I can’t really say personally.

Good Luck!

also there is a error icon in the upper left corner of the phone screen.

You may want to double check if TCP enabled and binded in sip.conf ; and in Avaya 9650 , set SIP Proxy to your asterisk .

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 …

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!

cheers,
Eranga.

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.

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)

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

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 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.

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 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.