Cisco VoIP Features

I needed to put together a list of available call manager features along with a high level description of what they are/do. I couldn’t find something that met my needs so I created it and thought I would pass it along in case it would help anyone else out. Obviously this not a comprehensive list and is customized for my purposes but it does cover many of the basic features.

  • Voicemail
    User Voicemail – Users can be assigned a voicemail box to which callers will leave messages when the phone is not answered. Voicemail greetings can be customized to use a system generated pronunciation of the users name, a recorded user’s name, or a fully recorded custom greeting. When a message is left by a caller the message waiting indicator on the phone will be lit.
    General Mailbox – A general mailbox allows a voice mailbox to be shared among a group of users for common access. Each member of the group has access to retrieve messages from the voicemail box.
  • Auto Attendant
    An auto attendant can be used to present callers with prerecorded options for selecting which departments or persons they would like to reach. Auto attendant options can allow a caller to directly dial a desired extension, transfer from the auto attendant menu to a preconfigured extension, or be setup in a hierarchical structure of options to guide the caller to a specific selection.
  • Jabber
    Jabber is a multi-featured communications client that enables several different ways of communicating with others.
    Presence – A user’s current status can be displayed on other Jabber clients. Presence states can be based off of a user’s calendar, idle time, WebEx meeting attendance, phone use and other custom messages.
    Chat –Jabber serves as an instant message client to facilitate chat sessions between two or more users
    Desktop Video – Users with webcam installed on their PCs can use Jabber to launch video calls between themselves and other video enabled users.
  • Conferencing
    Users on a phone call can set up ad-hoc conferences with up to 8 participants without having to use a dedicated conference number. When two users are on a call together, they can use the “Conf” softkey to dial and add other participants. 
  • Speed Dials
    Per location speed dials can be configured by Infrastructure Support.
    Individual Speed Dials (Abbreviated Dial) can be configured by the end user. 
  • Call Park/Directed Call Park
    Call Park – When on an active call, a user selects Call Park and a number is chosen by the phone system for the park. The park number is displayed on the phone for later retrieval. The call can be retrieved from the park, by a user dialing the call park number from any phone within the location.
    Directed Call Park – When on an active call, a user selects Directed Call Park and dials a pre-configured number at which to park the call for later retrieval. The call is retrieved by a user dialing the pre-configured number from any phone within the location.

  • Call Pickup Groups
    Call Pickup – Phones are placed into call pickup groups by Infrastructure Support. When a call comes inbound to a phone within the pickup group, any other phone within that group can pick up the ringing call by pressing the “Call Pickup” softkey and then the “Answer” softkey.
    Group Pickup – Phones are placed into call pickup groups by Infrastructure Support. When a call comes inbound to a phone in a different call pickup group, a user can answer the ringing call by pressing the “Group Pickup” softkey, dialing a preconfigured group number and pressing the “Answer” softkey.
  • Single Number Reach
    End users can configure an alternate number, such as a cell phone that also rings when calls come inbound to their primary phone. The ringing of the alternate number can be scheduled for time of day and access lists can be configured to only allow or deny calls from certain numbers.
  • Device Mobility
    When single number reach is enabled with a mobile phone, a user can start a call on their VoIP phone and seamlessly transfer the call to their mobile phone. Inversely, when a comes inbound to their VoIP phone and the user answers it by picking it up on their mobile phone, the user can seamlessly transfer the call back to their VoIP phone.

 

 

Echo Analysis on Analog Gateways

Here is a really great article about Echo Cancellation etc. for Cisco Gateways.

http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.htmlImage

MeetMe UCCX Script

After our migration to a Cisco VoIP system we decided to save some money and migrate our conferencing off of a paid conference bridge to use MeetMe exclusively for our conferences. Of course MeetMe is what is and although it is very easy to operate and convenient it isn’t exactly loaded in the bells and whistles department.  It has no audio front end or passcode requirement capabilities and provides no way of holding a call if a conference hasn’t been started by the organizer.

There are several solutions floating around the interwebs to fix some of the limitations. Some use unity and some utilize UCCX for an audio front end. I have taken several of the ideas that use UCCX and tweaked/compiled those for the solution posted here.

This UCCX script allows for up to 30 MeetMe numbers with a passcode validation for those who join. It will check to see if the meeting has been started and place the caller on hold if it has not. If you configure an audio stream for it you can play MoH for those on hold. Once the conference is started by the organizer each participant will be taken off hold and placed into conference.

Step-by-Step the script will work like this:
1.       Users will call the DID set up for external conference users
2.       The DID is pointed via a CTI route point  to the UCCX trigger of the MeetMe application
3.       The UCCX application begins the script, welcomes the caller and prompts for the conference ID
4.       If the conference ID is incorrect it will prompt 3 times before ending the call
5.       After the conference ID is verified it will prompt for the corresponding passcode
6.       If the passcode is incorrect it will prompt 3 time before ending the call
7.       Once the passcode is verified it will attempt to place the caller into the conference
8.       If the conference is not active it will play an audio prompt telling the caller to wait and play MoH
9.       Every 10 seconds it loop through the script to check if the conference is active
10.   Once the conference is active it will place the caller in conference and the script will end

Each of the Meeting ID’s and Passwords are set as parameters and can be set from the Application Management page.

The port group should be set to the numeric ID of the UCCX Telephony Call Control Group you want to use.

MeetMe

The Script and audio files are – Here

It is clunky and long but functional.

The audio files are to be placed in en-US\MeetMe\

This solution is designed on CUCM 8.6 and UCCX 8.5

A shoutout to WIKLUNDS who gave me the foundation for this script HERE