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Voice Mail (Audix ) calls not routing to users Mail Box

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TriniTech87

Technical User
Jul 4, 2016
70
TT

For the past few weeks we have been having routing issues with our Voice Mail system
below is a list trace that we conducted



time data

10:25:00 TRACE STARTED 09/29/2016 CM Release String cold-03.0.124.0-21754
10:25:05 active station 2368 cid 0x23bd
10:25:05 G711MU ss:eek:ff ps:20
rgn:4 [192.168.234.108]:2454
rgn:4 [192.168.234.11]:2064
10:25:07 dial 2370
10:25:07 ring station 2370 cid 0x23bd
10:25:37 no answer station 2370 cid 0x23bd
10:25:41 coverage-path 5 point 1 cid 0x23bd
10:25:41 call-forwarding 63900
10:25:41 term trunk-group 3 cid 0x23bd
10:25:41 call-forwarding 63900
10:25:41 route-pattern 3 preference 1 location 1/ALL cid 0x23bd
10:25:41 seize trunk-group 3 member 2 cid 0x23bd
10:25:41 Calling Number & Name NO-CPNumber NO-CPName



LIST TRACE

time data
10:25:41 SIP>INVITE sip:3900@sip.ngc.corp SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300
10:25:41 Setup digits 3900
10:25:41 Calling Number & Name 2368 Telecom Dept.
10:25:41 Proceed trunk-group 3 member 2 cid 0x23bd
10:25:41 SIP>ACK sip:3900@sip.ngc.corp SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300
10:25:41 SIP>INVITE sip:3900@sip.ngc.corp SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300
10:25:41 SIP>ACK sip:192.168.201.15:6061;transport=tls;gsid=80aa4ae7
10:25:41 SIP>-6585-4601-a202-56f69a730000 SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300
10:25:41 active trunk-group 3 member 2 cid 0x23bd
10:25:41 G711MU ss:eek:ff ps:20
rgn:1 [192.168.201.15]:8000
rgn:4 [192.168.234.11]:2068



LIST TRACE

time data
10:25:41 xoip options: fax:T38 modem:eek:ff tty:eek:ff uid:0x5002c
xoip ip: [192.168.234.11]:2068
10:25:41 SIP>INVITE sip:192.168.201.15:6061;transport=tls;gsid=80aa4
10:25:41 SIP>ae7-6585-4601-a202-56f69a730000 SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300
10:25:41 G711MU ss:eek:ff ps:20
rgn:1 [192.168.201.15]:8000
rgn:4 [192.168.234.11]:2068
10:25:41 xoip options: fax:T38 modem:eek:ff tty:eek:ff uid:0x5002c
xoip ip: [192.168.234.11]:2068
10:25:41 G711MU ss:eek:ff ps:20
rgn:4 [192.168.234.108]:2454
rgn:1 [192.168.201.15]:8000
10:25:41 SIP>ACK sip:192.168.201.15:6061;transport=tls;gsid=80aa4ae7
10:25:41 SIP>-6585-4601-a202-56f69a730000 SIP/2.0
10:25:41 Call-ID: 80aa4ae76585e61e5256f69a7300



LIST TRACE

time data
10:25:41 G711MU ss:eek:ff ps:20
rgn:1 [192.168.201.15]:8000
rgn:4 [192.168.234.108]:2454
10:27:03 SIP>BYE sip:192.168.201.15:6061;transport=tls;gsid=80aa4ae7
10:27:03 SIP>-6585-4601-a202-56f69a730000 SIP/2.0
10:27:03 Call-ID: 80aa4ae76585e61e5256f69a7300
10:27:03 idle station 2368 cid 0x23bd
10:31:30 TRACE COMPLETE station 2368 cid 0x0
 
What you mean bro ?
My explanation was poor posted it in a rush
 
What's the problem with the call you've traced? What happened? :)
 
It just want back to the audix home prompt (basically like I called my audix) rather than routing to the users Mail Box. so basically no one can send messages. It just goes to the aduix user menu
 
So on trunk group 3 , whatever you have the numbering set to "private or public" do you have the station details in the related unknown numbering table ?

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
When did it break and when did it work?

Something changed...

And if your voicemail is Audix, not IA770 or CMM, what's the integration? Usually it was a CLAN to send AUDIX the called/calling info and a hunt group of analog ports for the speech path, or maybe a DS1 integration.

What kind of AUDIX do you have and how are calls supposed to get there?
 
From what I gather our system uses a SIP trunk to go from CM to SM it's hunt group is empty as it uses virtual extensions
We don't utilize CLAN's again
 
There's no AUDIX I know of that uses SM and SIP. Maybe Communication Manager Messaging or Aura Messaging.

What's on the other end of the SIP trunk? A voicemail directly connected or another PBX that goes to an older AUDIX?
 
Aura Messaging but I don't have much info on it
The trunk a SIP based one and that's as much info I have on it to be honest
 
You need to get into both...

Is the pilot number right? What's AAM looking for? PAI or diversion headers for which user's greeting to answer with? Is your CM sending that, etc.

Basically, double check an integration note and see if it all lines up.
 
So it was the public unknown table (I did not put it back when i installed a PRI circuit )
So would need to add backs some numbers to the table
 
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