Get-CsVoiceRoute - View PstnGatewayLists

Always loosing this command, and takes me a few tries to find it again for what I'm looking for.

In the CSCP when looking at the associated trunks of a voice route, the trunk names are "truncated" and not fully displayed.

The command "Get-CsVoiceRoute" will display the full list of gateways that have been associated..

The above images is showing three (3) PstnGateways configured for the "Emergency" voice route

Call Dropping at 30 seconds when placed on hold

This has been done on many other blogs around the place, but I ALWAYS seem to get the first step wrong.

When trying to change the following values:
-RTCPActiveCalls
-RTCPCallsOnHold
-EnableSessionTimer

First step is "New-CsTrunkConfiguration", NOT "Set-CsTrunkConfiguration"

Trying to run the "Set-CsTrunkConfiguration" first will result in errors like below

How to extract and analyze the errors from the hybrid migration report using PowerShell

Found this article, whilst trying to figure out what mailbox items were causing issues with a migration from on-premises to Office 365.

How to extract and analyze the errors from the hybrid migration report using PowerShell

Command Examples:
Get-MoveRequestStatistics -IncludeReport -Diagnostic verbose | Export-CliXml .\move_report.xml

$r=Import-Clixml .\move_report.xml

$i=0;$r.report.Failures | foreach { $_ | Select-Object @{name="index";expression={$i}},failuretype,Message,timestamp;$i++} | ft 

$r.Report.Failures[4]

$r.Report.Failures| select -last 2

MoH - Skype, AudioCodes and an SIP Carrier

Troubleshooting an MoH issue today, where a SfB client would places a PSTN caller on hold, and the PSTN caller would get no MoH.  

SBC connection to PSTN was via a SIP trunk.

What I found was:
When call placed on hold from SfB client the INVITE had sdp of a=sendonly
This was received by the SBC and the AudioCodes "Remote Hold Format" on the IP Profile associated with the Skype side was set to "Transparent"

The "Remote Hold Format" on the ITSP side was also set to "Transparent"

In this instance the INVITE was forward to the carrier with an a=sendonly attribute.

But the ITSP would reply with a "200 OK" but the attribute was set to a=inactive.  This effectively shutdown the media path and the ITSP would not get the MoH stream from the SBC.

The Fix:
After reading the manual for the AudioCodes SBC and seeing what all the values in the "Remote Hold Format"

• Transparent = (Default) Device forwards SDP as is.
• Send Only = Device sends SDP with 'a=sendonly'.
• Send Only Zero ip = Device sends SDP with 'a=sendonly' and 'c=0.0.0.0'.
• Inactive = Device sends SDP with 'a=inactive'.
• Inactive Zero ip = Device sends SDP with 'a=inactive' and 'c=0.0.0.0'.
• Not Supported = This option can be used when the remote side does not support call hold. The device terminates call hold requests received on the leg interfacing with the initiator of the call hold, and replies to this initiator with a SIP 200 OK response. However, call retrieve (resume) requests received from the initiator are forwarded to the remote side. The device can play a held tone to the held party if the 'SBC Play Held Tone' parameter is set to Yes.
• Hold and Retrieve Not Supported = This option can be used when the remote side does not support call hold and retrieve (resume). The device terminates call hold and call retrieve requests received on the leg interfacing with the initiator of the call hold/retrieve, and replies to this initiator with a SIP 200 OK response. Therefore, the device does not forward call hold and/or retrieve requests to the remote side.

When the "Remote Hold Format" on the IP Profile associated with the ITSP was change to Hold and Retrieve Not Supported, the behavior of the SBC was changed to that once it received the INVITE with a=sendonly, it did NOT forward to the ITSP, so the media path was never altered from SBC to ITSP, and MoH was then played to the PSTN caller..

Test Phone Numbers

Was troubleshooting a DTMF issue today, and found some numbers from internode, that I think will be handy in the future.

Test Number	Functionality
08 7127 1201	Echo Test: plays your own voice back to you.
08 7127 1202	Music: plays fifteen minutes of Radio Paradise.
08 7127 1203	CLID Presentation Check: reads back your Calling Line ID to you, if it's turned on.
08 7127 1204	Talking Clock: reads back the current time to you.
08 7127 1205	DTMF Tone Test: reads back the digit pressed on your telephone keypad.

https://www.internode.on.net/support/faq/phone_and_voip/nodephone/feature_codes/

Tesing a fax number:
https://app.hellofax.com/

Mitel MiVoice Connect Contact Centre - Redundancy

EICC via Mitel MiVoice Connect with primary ECC server running

EICC via Mitel MiVoice Connect with redundant ECC server running

Complete the following steps to place appropriate configuration files on the HQ server. 

This ensures that if the system fails over, HQ is configured to run the contact center properly:

1. On the contact center server, navigate to 
location>\nginx\conf, and copy the following files:

• nginx_ecc.template
• readme.txt
• Redundancy.bat
• Redundancy_config.rb

2. On the HQ server, navigate to 
\nginx\conf\more_conf

directory, and paste the files you copied in step 1.

3. Double-click the Redundancy.bat file.
When the Windows console loads, follow the instructions on the console, and enter the IP addresses of the headquarters server, primary contact center server, and secondary contact center server.

4. Restart the ShoreTel-DirectorProxy service as follows:

a. In the Windows Start menu, click Control Panel > Administrative Tools > Services.
b. Select the ShoreTel-DirectorProxy service, and click Restart.

Cisco UC Tools

Some great tools for Cisco UC deployments

DRS Backup Decrypter

UCOS Password Decrypter

IFX Ontape Browser

https://www.adhdtech.com/uctools.html