Cisco 7940 phones on IPO 8.1

[Phoneguy15]

I don't know how I keep ending up in these situations.

Now I am tasked with getting Cisco 7940 telephone sets working with IPO 8.1
I have managed to get the firmware changed over from the Cisco Skinny to SIP.
I am not able to get them registered to the IPO.

The only time they will show registered (in Monitor) is when I turn off "Force Authorization" on the Extension Tab.
They will then register and I can place calls from the Cisco set but only have one way talkpath (Cisco can hear both ways but the other end cannot hear the Cisco).
If I turn "Force Authorization" on the set will never register.
There are not that many settings to change on the phone for registration (or on the IPO for that matter).

Here's hoping that someone out there has gotten this working and can point me in the right direction.

Thanks in advance

 

[CarGoSki]

is the gateway address the same address as your other ip phones that do work?

 

[amriddle01]

I don't know how I keep ending up in these situations

Stop agreeing to do them, we will not put in a system if this is a requirement...let someone else have the headache and ongoing issues

 

[Phoneguy15]

@CarGoSki, I have an Avaya 4620 connected to a test system here in the office and it is connected just fine to the same gateway.

@amriddle, you are absolutely correct. I need to learn to say "not just no, but hell no".

 

[Phoneguy15]

Here are the settings I have in place for the Cisco 7940 to register with:

Under "SIP Configuration"-

Preferred Codec: g711ulaw
Register with Proxy: Yes
Register Expires: 3600
Backup Proxy: 192.168.42.1
Backup Proxy Port: 5060
Emergency Proxy: 192.168.42.1
Emergency Proxy Port: 5060
Outbound Proxy: 192.168.42.1
Outbound Proxy Port: 5060
NAT: No

Under "SIP Config/Line 1"-
Name- 166 (is just a label?)
Short Name- 166 (is just a label?)
Authentication Name: SIP Test 2 (is User| User| Name)
Authentication Password- 1234 (is User| User| Telephony| Supervisor Settings| Login Code)
Display Name- 166 (is just a label?)
Proxy Address- 192.168.42.1
Proxy Port- 5060

 

[Phoneguy15]

BTW, here is the message that is showing up in Monitor:

104322130mS SIP Reg/Opt Tx: phone
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.42.129:5060;branch=z9hG4bK32f81e52
From: <sip:166@192.168.42.1>;tag=000f34d6f85903b85c75100f-6ba713b1
To: <sip:166@192.168.42.1>;tag=219f9a1a83c0e243
Call-ID: 000f34d6-f8590003-1f0fc1e3-34f5936c@192.168.42.129
CSeq: 574 REGISTER
User-Agent: IP Office 8.0 (18)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH

http://WWW-Authenticate:

Digest nonce="5562a4ec6ad12aacd824",realm="ipoffice",algorithm=MD5
Supported: timer
Content-Length: 0

 

[DrDrayAvaya]

Hi PhoneGuy15, i'm nearly diving into similar situation. client has 250 X 7940's over 18 sites, and how odd is this,.. they own the handsets outright and are leasing the system and voicemail server (call manager/ zeacom) ??? odd.

Presume G.711u law OK for system where you are, if system default then codec would be auto decided anyhow ay unless forced. (we usually nail it down to G.711A Law down the bottom of the world where we are NZ, system and phones, and SIP trunks)

What is the other end you're talking to? if it's Avaya IP phone, is the tick box for "Direct Media Path" ticked? if there's codec mismatch or something funky going on, if you untick this, should force VCM channel to stay in use during call. may be some mismatch with codecs between the other end phone you're testing to and the Cisco phone?

RTP range shouldn't be an issue, or you'd get no speech at all...

Check system status VCM use, and copy monitor trace of one of the test calls, see what RTP ports are in use, and what codec each end is using.

You'll have all the reg stuff right if you can make calls in a basic form.

Cheers,

Chris