Running Nortel i2004 phones on Asterisk

[hrpty]

Hello.  After nearly 2 weeks of trying to research this and try different configurations I finally have to break down and ask for help.

I am new to Asterisk and I am trying to setup and test run some i2004 phones that I purchased used.  I am now back to square one with a completely fresh install of AsteriskNOW 1.7.1 (Asterisk 1.6.2.11 and FreePBX 2.7.0.0).

I have read the other posts about configuring the i2004 phones for Asterisk however I am still getting the same results on the display of the i2004 phones every time.  ĻServer UnreachableĻ

I have these i2004s connected to a Baystack 460 running on PoE.  I have pinged the phone and it comes back fine.  I have pinged the server and it comes back fine.

Any help would be much appreciated.

[Michael McNamara]

Hi hrpty and welcome to the forums!

The i2004 and i2002 are almost identical so I would advise you check out this article I wrote a few years back.

http://blog.michaelfmcnamara.com/2009/02/asterisk-with-nortel-i2002-ip-phones/

You need to use the UNIStim channel driver (hopefully it's still in Asterisk) since these IP phones are not capable of SIP. You also didn't mentioned that you configured the IP phones with the IP address and port of the Asterisk server. You should be able to run Asterisk in debug mode (or perform a packet capture from the server) to see the IP phones attempting to connect.

Good Luck!

Cheers!

[hrpty]

Ok. I have read this thoroughly and everywhere is stated that Asterisk v1.6 and up come with Unistim in the Asterisk tree source.  Perhaps there is something else that needs to be installed with this that I have not found?

I have reviewed the postings here regarding the settings for the phone itself. It is configured to my Asterisk serverīs IP address and port 5000.  Still a no go.

In any case, I get the impression from what Im reading that I still need to edit the unistim configuration file.  I have not, however, been able to either locate or open this file to edit it.  Forgive me, I am very new to Asterisk and Linux.

Your help is much appreciated,
Matt

[hrpty]

It appears that there was a problem with the distro of AsteriskNow I was using, missing unistim files.  In any case, I went with a fresh re-install of TrixboxCE, edited the unistim.conf (just like the one you posted), created the extensions, connected two i2004s on PoE and voila, they connected and registered with Asterisk.

Now the problem is this.  I am running these two i2004 test phones as extensions and 2 LAN terminals with X-Lite extensions.  The X-lite will dial and communicate fine between each other and the i2004s will ring the X-Lite extensions.  However, the i2004s will not ring each other.

Any ideas on that one?

[Michael McNamara]

I don't recall having any such issues when I tested it sometime ago. I can't really say myself perhaps someone else will be able to tell. Short of that I would look to put Asterisk into debug and see what the logs say when you try to dial the handset. Are you selecting the line and then dialing the number and selecting dial again?

[Caraculo]

Hi.

I have a Nortel i2002 and I am posting this here since the phone is similar to the i2004 when it comes to its technology.

I am also trying to make it work through an asterisk 1.6 environment using unistim.conf to be able to use the i2002 UNIStim capabilities.

My Asterisk environment works fine when running with a Cisco 7960 SIP enabled phone but I cannot make it work with the Nortel i2002. I can actually call local numbers but no external calls. I have read McNamara's comments and his blog but without being able to find out what is going wrong. I also searched the web but also with no success.

One thing that I don't understand is that my "exten=>" doesn't contain any trunk information as when using SIP. So how does the Nortel phone ans Asterisk know how to reach the "outside" world?

Here are the Asterisk-log lines from call and my conf files...

Asterisk log
[Jan 16 00:08:35] NOTICE[12082]: chan_unistim.c:3582 parsing: 192.168.1.7 Warning : received a retransmitted packet : #0x0000 (we are at #0x0001)
    -- Device 'i2002' successfuly registered
    -- Added extension '208' priority 1 to default (0x603098)
    -- Starting switch on '208@i2002-0' to 65555555
  == Starting USTM/208@i2002-0 at default,55555555,1 failed so falling back to exten 's'
  == Starting USTM/208@i2002-0 at default,s,1 still failed so falling back to context 'default'


unistim.conf
[general]
port=5000
autoprovisioning=no

[i2002]
context=default
device=0016 xxxx xxxx
rtp_port=10000
rtp_method=3
status_method=0
dateformat=0
timeformat=2
country=dk
callerid="Customer Support" <208>
callhistory=1
linelabel="208"
line=>208

extensions.conf
exten => 208,1,Dial(USTM/208@i2002)

Working extensions.conf used with Cisco SIP ; not used when testing the Nortel i2002
[CallingRule_208]
exten => _X.,1,Dial(SIP/${EXTEN}@musimi,40,r)

users.conf
[musimi]
context = DID_######
host = musimi.dk
trunkname = Musimi
username = #####
secret = xxxxxxxx
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
insecure = port,invite
disallow = all
allow = g729,ulaw,alaw

[208]
username = 208
transfer = yes
mailbox = 208
host = dynamic
...
dtmfmode = rfc2833
disallow = all
allow = g729,ulaw,alaw

[Michael McNamara]

Hi Caraculo and welcome to the forums!

I've never had the opportunity to test calls outside a single Asterisk server, although I believe there was a IAX2 test trunk (years back now - probably gone now) that I was able to setup and test calls through. I've also been successful in establishing calls between an 1120e (SIP) and an i2004 (H323) so Asterisk must be normalizing the signaling (H323 <-> SIP) and feeding the RTSP media G.711 packets between the endpoints.

Obvious question here... have you tried to setup the Extern string just like any other SIP extension, substituting the proper extension definition?

I would think it's just an Asterisk configuration issue... unfortunately I'm not an Asterisk expert although I have been playing with it more recently.

Cheers!

[hrpty]

I have these Nortel i2004s (24 of them) powered up to a Nortel Baystack PoE switch and communicating with Asterisk v2.6, (Trixbox v2.8.0.4) on a Dell Poweredge 1750 server...

Anyhoo... I can get them to ring, display anything I want. All the ring tones work fine, gets dial tone, But I STILL can not get any audio whatsoever in either direction - calling internally extension to extension. I have also tested this with X-Lite 4 calling out to a Nortel set and receiving in to a Nortel set. Same results as Nortel to Nortel, dial tone, it rings, says call is connected but no audio.

So, the question is, could this be the firmware on the Nortel sets??  Does a firmware update even matter?  And, how do you do a firmware update without a BCM?

[Michael McNamara]

So, the question is, could this be the firmware on the Nortel sets??  Does a firmware update even matter?  And, how do you do a firmware update without a BCM?

I though think the firmware wouldn't be an issue with just making calls.

There is no way to update the i2002/i2004 without a CS1000 or BCM or similar as they utilize the UTFTP process.

So you've made some progress from your initial post... care to share with the folks here how you got this far?

[hrpty]

The original problem was with the newer version Distro of Asterisk v2.8. I couldnīt figure out unistim.conf, because aparently it didnīt come with this preloaded. Even after trying to install it via Yum, I was not successful. So, I switched to an older Distro - Trixbox CE (v2. and it was easy to run. I simply edited unistim.conf with similar settings as those you have listed on the forum, reload the config files and then added the extensions in the Web GUI and update settings. After that, the phones register with Asterisk and display all options on the screen in working function, great! They ring each other by extension number, great!

Then the problems come into action. All ring/dial tones work, but No voice audio on any end and they donīt dial out either. Also, Asterisk is not showing the phones online or even being in the queue. I have tested this with DHCP and with Static IP with the same results and wonder if NAT could be the problem? I have read up on setting the Nortels up in a VPN. This may be the next step...

Thoughts anyone?

[Michael McNamara]

Are you pressing the line key? I believe you need to press the line key to place/answer a call.

[hrpty]

If I understand correctly...

The line key is the green of the six soft keys and opens up a dial tone.  Simply picking up the handset does exactly the same thing.  There is 1 of 4 feature keys below the display, one of which is marked for DIAL, pressing that does the same thing also!  After doing any of the above three, that same lower softkey then changes to say CALL, which I do, and then press it. 

That does ring extension to extension internally. And, when dialing out, it says "is on-line" and the call timer runs, but no voice/audio either case - internal or external.  Itīs baffling.

[Michael McNamara]

I just setup an i2002 tonight really quick using my recent AsteriskNow installation, details regarding the configuration can be found here. I created the file /etc/asterisk/unistim.conf, I added a custom extension definition in FreePBX, and I restarted Asterisk (service restart asterisk).

Initially I was unable to establish a voice path between either Asterisk or another extension. Once I added the following three lines to my unistm.conf file under the extension definition and after restarting asterisk everything starting working.


rtp_port=10000              ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
rtp_method=3                ; If you don't have sound, you can try 1, 2 or 3, default = 0
status_method=0             ; If you don't see status text, try 1, default = 0


I retrieved the information above from my original blog post on Asterisk back in February of 2009.

Hopefully this helps.

Good Luck!