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SAC - Remote not working

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guycable2

MIS
Apr 16, 2004
363
0
0
US
Strange issue here. This worked for a while and then stopped.

Setup
CM 6.3
AAM 6.2
SM 6.3
SIP 9611 phones

We have several sites connected to this CM. The sites have main numbers that ring to a coverage path that contains a cover answer group for their site during the day and AAM at night. When they go home for the night the users press an auto dial button that is programmed with the remote SAC FAC (*154) plus the main number, when they come in in the morning they press another button programmed with the remote SAC cancel FAC (#154) plus the Main Number. The main number is programmed as a 500 xport extension. Cover Path 1 is set to DND/SAC/Goto Cover no for both inside and outside. Cover path 2 is set to the Voicemail coverage path so DNC/SAC/Goto Cover is y for both.

Now this stopped working sometime in the last week. When selecting the enable button (*154 + main number) the user receives the reorder tone. A list trace station shows a denial event 1461: User cverage path missing.

I added both coverage paths to my phone and the system won't allow me to press SAC either. What parameter could have changed that would affect this feature?

Thanks

 
do a list history maybe someone made a system change recently that affected this

 
indeed does the cp still exsist?

APSS (SME)
ACSS (SME)
ACIS (UC)
 
Yes the CP still exists.

I have gone back through the history and have not found any changes yet but we are building a log of phones so it could be lost in that list somewhere. I don't know of any system parameters that have been changed though.
 
very weird that you can not enable send all calls, is this for everyone or just you?

 
so somone has naffed up the cover path COR or the like , so from
a set with console permmissions can you diaL *154 and your cell phone then call and it works ???

if not place a call including *154 list trace it and show us the results

APSS (SME)
ACSS (SME)
ACIS (UC)
 
Failed to my cell phone as well.

Here is the trace. I changed the phone numbers to bogus ones.

09:08:05 TRACE STARTED 10/16/2014 CM Release String cold-03.0.124.0-21291
09:08:11 SIP<INVITE sip:17003991234@sip.abc.com;avaya-cm-fnu=of
09:08:11 SIP<f-hook SIP/2.0
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP>SIP/2.0 183 Session Progress
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP>PUBLISH sip:17003991234@sip.abc.com SIP/2.0
09:08:11 Call-ID: 808cbc431460e41fe45542c783500
09:08:11 active station 17003991234 cid 0x1460
09:08:11 SIP<SIP/2.0 200 OK
09:08:11 Call-ID: 808cbc431460e41fe45542c783500
09:08:11 SIP>SIP/2.0 484 Address Incomplete
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP<INVITE sip:*154@sip.abc.com SIP/2.0
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP>SIP/2.0 100 Trying
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP>SIP/2.0 200 OK
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 G711MU ss:eek:ff ps:20
rgn:1 [172.24.123.183]:5020
rgn:1 [10.24.25.71]:2074
09:08:11 xoip options: fax:Relay modem:eek:ff tty:US uid:0x500bb
xoip ip: [10.24.25.71]:2074
09:08:11 SIP<ACK sip:172.24.21.105;transport=tcp SIP/2.0
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:16 dial *15417003995678
09:08:16 term station 17003995678 cid 0x1460
09:08:16 denial event 1461: User cverage path missing D1=0xa017 D2=0x1460
09:08:19 SIP<BYE sip:172.24.21.105;transport=tcp SIP/2.0
09:08:19 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:19 SIP>SIP/2.0 200 OK
09:08:19 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:19 SIP>PUBLISH sip:17003991234@sip.abc.com SIP/2.0
09:08:19 Call-ID: 804081481460e41446542c783500
09:08:19 idle station 17003991234 cid 0x1460
 
so this bit means not enough digits are being sent
9:08:11 SIP<SIP/2.0 200 OK
09:08:11 Call-ID: 808cbc431460e41fe45542c783500
09:08:11 SIP>SIP/2.0 484 Address Incomplete
09:08:11 Call-ID: 4b_ac1657e29f8b3b95f0136ea_I@172.24.123.183
09:08:11 SIP<INVITE sip:*154@sip.abc.com

the rest is the network not accepting the digits and interestingly a mixture of the number you are dialling

APSS (SME)
ACSS (SME)
ACIS (UC)
 
So I took the second Coverage Path off the phone and set the 1st coverage path to DNC/SAC/Goto Cover is y for both inside and outside and the feature works.

So for some reason it's not recognizing that there is a second coverage path on the phone.

I opened an Avaya case too.
 
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