UC Recording

http://ciscoshizzle.blogspot.com/2013/09/call-recording-configuration-in-cucm.html

UCCX Scripts Scenario

Scenario 001: Check holiday from XML and do different function on holidays.

It can be done!

Scenario 002: Depending on IVR calling number Script will do a different function.
It can be done!

Scenario 003: Changing default prompt by an end user or manager for holiday or emergency.
I know it can be done.But I have not done anything like this yet.Will work on this when I get time.
ref: Click here




For any type of UCCX script help, drop me a line.
Remember "If You're Not Paying For It, You Become The Product"

UC QoS

ref

https://thevoiceman.wordpress.com/2011/10/19/configuring-voice-qos/

DIal-Peer Binding issue

Cisco UC gotcha is "Unable to add dial peer binding while there is an active session".


My solution:

For inbound call, we rerouted all call to different SIP circuit

First, you can get a good overview of active calls with the command 

#show call active voice (It may say no active call but look for the call legs:##)

#Show call active voice compact (It will show list of active call session on CUBE and number associated with it)

"ANS" Meaning= This number initiated call(calling Number)

"ORG" Meaning= This number received call(Called Number)

#sh voip rtp connections (It will show remote end and end point ip address for active session)

Note***If you only get ITSP IP address as remote and Endpoint most likely those calls are hung call on CUBE.

The command to Clear hung calls:
"clear call voice causecode 1  called-number 91414XXXXXXX"
"clear call voice causecode 17 calling-number 91414XXXXXXX"

clear call voice causecode identifier{id identifier | media-inactive | calling-number number |called-number number}



what is cause-code:


Then we open a d ticket with Cisco on this issue. Open Cisco Recommendation was:

Below are the commands to forcefully shut down the sip service on CUBE.

Router# conf t
    Voice service voip
    Sip
    Call service stop forced
====
This will get rid of all sip sessions (existing and transient)
"Personally I do not agree with Cisco about force stop we should do graceful stop.Then again we were waiting for one call to finish to half an hour then called that guy and asked him to hang up. lol"

The way to monitor whether calls failover to secondary or a backup CUBE is to run show commands/debugs on the expected backup/secondary CUBE


“show sip sessions brief”
Show call active voice brief

Debug voip ccapi inout
Debug ccsip messages



Solution 1:

Solution 2:

ref: 
https://supportforums.cisco.com/t5/collaboration-voice-and-video/unable-to-stop-voice-call-processing-on-a-cisco-ios-gateway/ta-p/3133040
https://supportforums.cisco.com/t5/collaboration-voice-and-video/cube-sip-media-and-signalling-binding-to-an-interface/ba-p/3107153
http://ucpros.net/cisco-sip-gateway-configuration/

Never SCCP again

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VG 204/224

Things need to know for Analog and FAX gateway:

VG config using SCCP
*************************************
! stcapp ccm-group <groupnum> 
stcapp 
! voice-card 0 
! 
! voice service voip 
modem passthrough nse codec g711ulaw 
! sccp ccm <ipaddr> identifier <idnum1> 
sccp ccm <ipaddr> identifier <idnum2> 
sccp 
! sccp ccm group <groupnum> 
associate ccm <idnum1> priority 1 
associate ccm <idnum2> priority 2 
! 
! dial-peer voice <dialpeer1> pots 
service stcapp 
port 2/0 
! dial-peer voice <dialpeer2> pots 
service stcapp 
port 2/1 
! 
******************************************************

Cisco VG224 Configuration Example Click Here

For FXO line on cisco IOS router that remains connected even after the call ends

#Power denial-based supervisory disconnect

#Ground-start signalling disconnect

#Tone-based supervisory Disconnect

#disconnect-ack 


T-shoot:

The FXS ports on the VG224 Voice Gateway do not go off-hook, which results in a fast busy tone Click Here

Fax Pass-through Click here