Monday, 2 October 2017

T-Shoot Call drop.

Ring No answer incoming call from PSTN drop after exact 60 Sec!!. The internal call rings for 120 sec and then gets forwarded to voicemail as per CUCM configuration and I would like to achieve this for incoming PSTN call.Call from cell phone Verizon/ATT drops after 60 Sec. I also have noticed call from PSTN landline call kinda drop at 60 Sec then silent for around 2 Sec and start ringing again. Hah! Isn't it awesome!!!

Not sure, what causing them to send CANCEL and why exactly at 60Sec ???!

Cisco CNS blamed on ITSP which may make sense because ITSP sends CANCEL message first.And ITSP blaming on us. Huh!

Min-SE: 60

Session-Expires: 1800;refresher=uac

Max-Forwards: 68

Content-Length: 242

Content-Disposition: session; handling=required

Content-Type: application/sdp

I have opened a ticket with Cisco and also with my ITSP collected some log. Will share with you guys soon. Stay tuned.

https://tools.ietf.org/html/rfc4028

Saturday, 16 September 2017

UC Recording

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

Friday, 15 September 2017

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"

Thursday, 7 September 2017

UC QoS

ref

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

Friday, 1 September 2017

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/

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