Help with Nortel i2004 Phones

[ajph.voip]

Hello all,

I was just wondering if any of you know how to get Nortel i2004 phones working with PIAF.

So far I can only make calls out of them, but can't receive any.

Thanks in advance for any help.

I'm not an expert by any means, but really I'm looking forward to become an active participant in this community.

 

[tshif]

Check out this thread - maybe a place for you to get started.
http://pbxinaflash.com/community/threads/nortel-voip-phone-support.45/

 

[w1ve]

Hello all,

I was just wondering if any of you know how to get Nortel i2004 phones working with PIAF.

So far I can only make calls out of them, but can't receive any.

Thanks in advance for any help.

I'm not an expert by any means, but really I'm looking forward to become an active participant in this community.

Hi,

Tell me more. I have a bunch of i2004s running on my PIAF. If you can make calls out of them, your in good shape. There are a few critical sections in the unistim.conf file.

Make sure you have the line:
extension = line

followed directly by
line =>XXX

where XXX is the extension in the dial plan.

If the extension section is [Sample400]
and the extension is 400, you must create your i2004 Extension in FreePBX as a CUSTOM extension, and the format in the Dial string must be:
USTM/400@sample400

Other important questions:
- Is PBX and phones on same LAN?
- Is PBX and/or phones NAT'd?

Does PBX have dynamic IP? If so, you are screwed. Nortels require fixed ip.

If PBX is public IP, you can't have more than one phone behind NAT if phones in different LAN.

Also, when you say you can't call it, what happens? Don't rely on the call-progress tones... use the asterisk debugger.

Go to console and type asterisk -rvvv
You'll get lots of messages telling you what might be wrong.

Send more detail and we will get it working...

 

[ronw]

In your unistim.conf file, you should have:
line => xxx
where xxx = the extension number that you assigned in FreePBX.

 

[ajph.voip]

Got It Working

After reading and following you guys' advise and with the help of a friend I finally got these phones working.

My problem was in the unistim.conf file. in the "line => " part.
Also I did not know you had to create a custom extension. I was creating just a regular one.

Thanks for the advice you all, specially w1ve. Your info was very helpful.

 

[w1ve]

After reading and following you guys' advise and with the help of a friend I finally got these phones working.

My problem was in the unistim.conf file. in the "line => " part.
Also I did not know you had to create a custom extension. I was creating just a regular one.

Thanks for the advice you all, specially w1ve. Your info was very helpful.

awesome. Watch for forum for updates on "Asterisk Stickies" by aster1sk... He is working on a web page interface to configure i2004s and the other chan_unistim supported phones. No more messing with unistim.conf directly.

 

[webstar]

What I would like to know is how to fix it so the phone doesn't keep going goodbye when you do any feature codes. i can call extention to extention fine but outside of that it goes goodbye.

Including but not limited to *98 *97 7777 etc...

 

[w1ve]

What I would like to know is how to fix it so the phone doesn't keep going goodbye when you do any feature codes. i can call extention to extention fine but outside of that it goes goodbye.

Including but not limited to *98 *97 7777 etc...

Hmmm dunno... I put the feature codes I'm interested in on the soft buttons, and they work fine.

 

[webstar]

Hmmm dunno... I put the feature codes I'm interested in on the soft buttons, and they work fine.

Hmmm I posted this in another forum but this is what I get when I try.


Nortel i2002
I can call it from x-lite and call x-lite, I can call other phones and all the audio works.

The issue is nothing outside of extention numbers will work. When I diail them and press call they just hang up.

This is the lines in unistim.conf
device=000ae40920aa
dateformat=0
timeformat=0
line => 100
maintext0="Nortel i2002"
maintext2="Asterisk"
context=from-internal
extention=line
ringvolume=0
rtp_method=3


And this is what the CLI shows for..

NON-Working Call. To Sim incoming call (777)
-- Starting switch on '100@jason-0' to 777
== Starting USTM/100@jason-0 at default,777,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("USTM/100@jason-0", "vm-goodbye") in new stack
-- <USTM/100@jason-0> Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [s@default:2] Macro("USTM/100@jason-0", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("USTM/100@jason-0", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("USTM/100@jason-0", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("USTM/100@jason-0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("USTM/100@jason-0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("USTM/100@jason-0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("USTM/100@jason-0", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'USTM/100@jason-0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on 'USTM/100@jason-0'
USTM(100@jason-0) channel already destroyed


WORKING CALL
-- Starting switch on '100@jason-0' to 202
-- Executing [202@default:1] Macro("USTM/100@jason-0", "exten-vm,202,202") in new stack
-- Executing [s@macro-exten-vm:1] Macro("USTM/100@jason-0", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("USTM/100@jason-0", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("USTM/100@jason-0", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("USTM/100@jason-0", "1?Set(REALCALLERIDNUM=)") in new stack
-- Executing [s@macro-user-callerid:4] Set("USTM/100@jason-0", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("USTM/100@jason-0", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("USTM/100@jason-0", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("USTM/100@jason-0", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("USTM/100@jason-0", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("USTM/100@jason-0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("USTM/100@jason-0", "Using CallerID "" <>") in new stack
-- Executing [s@macro-exten-vm:2] Set("USTM/100@jason-0", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("USTM/100@jason-0", "VMBOX=202") in new stack
-- Executing [s@macro-exten-vm:4] Set("USTM/100@jason-0", "EXTTOCALL=202") in new stack
-- Executing [s@macro-exten-vm:5] Set("USTM/100@jason-0", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("USTM/100@jason-0", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("USTM/100@jason-0", "RT=20") in new stack
-- Executing [s@macro-exten-vm:8] Macro("USTM/100@jason-0", "record-enable,202,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("USTM/100@jason-0", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("USTM/100@jason-0", "recordingcheck,20090323-135038,1237830638.80") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20090323-135038,1237830638.80: Inbound recording not enabled
-- <USTM/100@jason-0>AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("USTM/100@jason-0", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("USTM/100@jason-0", "dial,20,Ttr,202") in new stack
-- Executing [s@macro-dial:1] GotoIf("USTM/100@jason-0", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("USTM/100@jason-0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 202 to extension map
-- dialparties.agi: Extension 202 cf is disabled
-- dialparties.agi: Extension 202 do not disturb is disabled
-- dialparties.agi: DbDel CALLTRACE/202 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 202
== Manager 'admin' logged off from 127.0.0.1
-- <USTM/100@jason-0>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("USTM/100@jason-0", "SIP/202,20,Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 202
-- SIP/202-b7d9eb98 is ringing




only extentions seem to work, but speeddials, features etc.. don't work. I can't even *97 to check voicemail.

 

[ronw]

This comment is in the sample setup in unistim.conf:
Beware ! only bookmark and softkey entries are allowed after line=>

You need to move your "line => 100" entry.

 

[webstar]

This comment is in the sample setup in unistim.conf:
Beware ! only bookmark and softkey entries are allowed after line=>

You need to move your "line => 100" entry.

oh yeah I feel stupid now.