Collaboration, Voice and Video Blogs

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Cisco just Announced their new Collaboration Toolbox resources portal. You can check it out here https://collabtoolbox.cisco.com/ It has a bunch of tools a partner can find helpful. But if your interested in learning which each Cisco tool is suited for ch... Read More

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Starting from Cisco Meeting Server 3.0.0 version you can get a trial demo through the Cisco Meeting Management. See below how to get this trial step by step and straightforward. https://ipdemystify.com/2020/09/27/how-to-get-a-trial-demo-for-cisco-meeting-... Read More

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Proud to announce a minor update of the unique resource and book about Cisco Meeting Server in amazon library, I pushed far the explanations to be simple as much possible with atypical chart call flow after a hard work. Enjoy studying and you will love it... Read More

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Cisco Meeting Server 3.2 supports the Blast Dial feature. With Blast Dial, you can add a predetermined list of participants to a space where you configure blast dial. When any participant dials in to the space, all the other participants are dialed out si... Read More

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Shortcut ActionCtrl+W Close Meeting CenterCtrl+T Transfer filesCtrl+Z Undo last actionCtrl+Y Redo last undone actionCtrl+Alt+F Display Font Formatting dialog boxCtrl+Alt+O Share presentation or documentCtrl+Alt+A Share applicationCtrl+Alt+D Share DesktopC... Read More

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Do you experience the following error on Cisco Unity Connection: Sometimes when you try to play or upload Audio Files for greetings in Unity Connection under the Call Handlers such as the System Call Handler Opening Greeting, in Standard/Closed/Holiday Gr... Read More

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This document describes the implementation of Clustering for Cisco Expressway Series Edge and Core from scratch.

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When an Ad hoc conference is triggered by an endpoint to escalate a point-to-point call into a three-way call. HQ-CUCM sends an API request via HTTPS to the Cisco Meeting Server to set up a conference. You can see from the event log of CMS that API user “... Read More

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You want in some scenarios different URIs or call-IDs and different access code are used by guest and host participants to join a meeting, and different rights or privileges, for example add or remove participants, mute audio or video, or when there are o... Read More

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Opus is an adaptive codec that provides better audio call quality than G.711/G.729 voice codecs and in a low bandwidth environment. The codec has a very low algorithm delay and is it is highly scalable in terms of audio bandwidth, bitrate, and complexity.... Read More

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Background: When deploying a multisite with centralized call processing across multiple countries, the digit manipulation can be complex, in this scenario US Site as HQ, GE Site in Germany and AU Site in Australia use different dial plan. We need to unifi... Read More

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Version 3.3 introduces the ability to schedule meetings and see upcoming meetings in web app. Web app users can schedule meetings, modify the scheduled meetings, and notify participants via email. Scheduler is a new component that enables scheduling meeti... Read More

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Translation Profile is used on Voice Gateway to perform Digit Manipulation on the Calling and Called Parties. The Translation Pattern is doing the same digit manipulation. The translation profile can be applied BEFORE the incoming dial peer is matched (if... Read More

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Dial Plan Call Routing Demystified Version 1.0: Cisco Collaboration Technologies Unified Communication Conferencing and Edge Kindle Edition Dial plan and Call Routing are the most important complex components for any Cisco Unified Communications system an... Read More

Latest Contents

ECE 12.5.1 - Agent Availability Check

Created by ryschule on 10-27-2021 01:40 PM

I am attempting to configure the Agent Availability check for a given Entry Point. I have: 1. Configured the Agent Availability option on the Entry Point to 'Required' 2. Configured the Agent Availability Check Mechanism to 'Absolute' 3. Configured... view more

SIP trunk call transfer issue

Created by aneesh.abraham on 10-27-2021 12:03 PM

Hello, I am facing call transfer issue over SIP trunk, we have CME connected to ISP with SIP trunk, user is not able to transfer incoming calls from SIP trunk to other extensions.Call flow -- PSTN-->SIP Trunk-->CME--&g... view more

8821 - Mark Call as SPAM

Created by chuck47172 on 10-27-2021 08:50 AM

Does anyone know if the 8800 series feature "mark call as spam" works on a wireless 8821?I don't think it does but if I'm wrong I could use some instructions on using the feature as I don't have an 8821 on hand.

SyslogSeverityMatchFound triggered by LogPartitionWaterMark

Created by drehstrom on 10-27-2021 08:43 AM

Hi fellas,I'm facing an issue with our CUCM regarding the triggered alerts. Within RTMT I see three alerts on a regular basis1. LogPartitionHighWatermarkExeeded2. LogPartitionLowWatermarkExeeded3. SyslogSeverityMatchFoundI can live with the first two as I... view more

Webex Control Hub & API Support

Created by ruatkin on 10-27-2021 08:24 AM

Greetings, does Control Hub have the ability to send troubleshooting metrics (latency, jitter, packet loss, etc) via an API/webhook?