denial event 1180: No user responding D1=0xe0011 D2=0x12

[Montero84]

Hello,

What does this error means?.. does somebody knows?

Thanks

 

[SF0751]

From the Maintenance Alarms manual:

Denial Event 1180
No user responding No user responding.
See Cause Value 18 on page 150.
Event Data 1: UID
Event Date 2: DIAG/LOC/CV

Cause Value 18 (0x12/0x920 - No user responding/No response from the remote device

The remote device/endpoint/PBX did not respond with an ALERTING/PROGRESS/CONNECT
indication within the time administered in the T303 or T310 timers Q.931 specification.
Cause Value 18 indicates high traffic conditions in the serving ISDN network or noisy conditions
on the T1/E1 span carrying the d-channel messaging. The noise is causing the loss of
messages being sent to the remote device. The remote device might also be unable to respond
to the incoming SETUP request.

This Cause Value has end-to-end significance and should always be passed back through the
network to the user.

Susan
"When the gods wish to punish us, they answer our prayers." - Oscar Wilde, An Ideal husband, 1893

 

[Montero84]

Hello Susan,

Thank you for the reply. I am trying to configure a sip between an Ascom Phone system and my avaya S8400. I have configured the sip trunk in the Ascom pbx and the sip trunk in the Avaya side. I can make calls from the Ascom phones to the Avaya phones but can't from Avaya to Ascom, I get the error message "no user responding, etc.." any idea on how to solve it?...

Thanks in advance!

 

[SquareWorld]

Did you ever resolve this issue?

 

[Montero84]

Hello SquareWorld

Nop, I am still working on it.. have any idea of what could be causing the error?

Thanks in advance!

 

[SquareWorld]

I'm afraid not. Unfortunately, my trace gives the same error as you. A Google search brought me here.

I do plan on buying some i75 Ascom phones (as recommended by Avaya) - but right now my problem similar to yours only with a SIP to Analog converter.

G450/S8300/SES to AudioCodes MP-124.
Inbound:
denial event 1180: No user responding D1=0xe0011 D2=0x12

I suspect my MP-124 is not registering correctly.

 

[Montero84]

Hello SquareWorld

I am still working on this project.. I have configured my SIP trunks, the route, the off-pbx-telephone options, etc.. but I am still receiving the same error message..

 

[jimbojimbo]

Most likely issue is the node-name you used for the remote end of the sip trunk. You can run an MST trace in CM to get better detail of the SIP failure. Make sure your node-name matches your host name and the domain is correctly set in either the signal group or IP-Network-Region form. MST trace will show the SIP messages so you should be able to get a clear picture.

 

[Montero84]

Hello Jimbojimbo

Thank you for the reply... here is what I got:

matching filter label <Inbound 5545>: SIP. domain .net: [Send Response ]

{connection: host=10.111.240.246 port=5061 protocol=TLS}

SIP/2.0 302 Loop detected by CM

From: "IT Portable" <sip:anonymous.invalid:5061>;tag=0d05cd2e176dd12c56484192a800

To: "5545" <sip:5545@domain .net>;tag=580D75A86EAD9409D0C1ED437926869D12181655636335

Call-ID: 0d05cd2e176dd12d56484192a800

CSeq: 1 INVITE

Via: SIP/2.0/TLS 10.111.240.246;psrrposn=2;received=10.111.240.246;branch=z9hG4bK0d05cd2e176dd12e56484192a800

Contact: sip:5545@domain .net;redirectingProxy=10.111.240.246

Content-Length: 0

I did check the domain and is correct, I also check the node-name and the ip-network.. and everything looks correct

Any idea?

Thanks in advance!

 

[SquareWorld]

This may be way out there in left-field - but if you go and do a trace in 'Trace Logger' does your sipTraceLog do you see odd extension registrations?

For example, I'm trying to register extension 2052. Yet when I do a trace I see all kinds of other SIP extensions that have nothing to do with the extension I programmed in:

Attempt to register 2052 comes up with these entries:

username="2057",realm="domain.domain",... 
...
username="2054",realm="domain.domain",...
...
username="2056",realm="domain.domain",...

2075, 2054, and 2056 are not programmed anywhere on my SES, MP-114 or Avaya systems. I'm not sure where they are coming from.

Yet, when I register with X-Lite (free SIP softphone) the extension 2052 works. (The 2057 etc disappear from the trace logs.) Additionally, if I use a Planet VIP-156PE Analog to SIP adapter everything works great.

 

[jimbojimbo]

If you don't set your Regisratar Gateway and Registrar Name to a specific extension you will get random extensions attempting to register as the gateway. Drove me nuts.

 

[jimbojimbo]

Did you set up the Host Map and Contact information on the SES to route calls? Run the command

ccstrace on file=/var/home/ftp/pub/ccstrace1.txt

on the SES server. Try the call then

ccstrace off

vi the file. You may need to map ^sips:xxxxx where xxxx is the digits needed to uniquely identify where the call must go then add the contact as <sip:$(user)@server.domain (or ip address).

 

[SquareWorld]

Thanks JimboJimbo, I'll try that.

Montero84, do you get inbound and outbound communications? (Are you using H.323 server as apart of the Ascom setup or just the phone as a straight up SIP phone?)

Perhaps you could try registering X-Lite with extension 5545. (To see if it works both-ways).