3. Configure Avaya Communication Manager

This section presents configuration steps for the Avaya Communication Manager. It is assumed that an appropriate license file and authentication file have been installed on the server, and that login and password credentials are available.

The configuration and verification operations illustrated in this section were all performed using the Communication Manager System Administration Terminal (SAT).

The procedures include the following areas:

• Verify the status CTI link for TSAPI service
• Disable RTP encryption
• Enable RTCP reporting
• Disable IP-IP Direct Audio (optional)
• Administer System Parameters Features

3.1. Verify the status of CTI link for TSAPI service

Log into the System Access Terminal (SAT) to enter the "status aesvcs cti-link" command. The link status should show no for maintenance busy (Mnt Busy), the Service State should indicate established and Version should be 6 or higher.

status aesvcs cti-link

                          AE SERVICES CTI LINK STATUS

CTI   Version  Mnt   AE Services      Service       Msgs     Msgs
Link           Busy  Server           State         Sent     Rcvd

1     7        no    aes              established   15       15

If the CTI link is not established, then follow instructions in chapter Administering Communication Manager for AE Services in document Application Enablement Services Administration and Maintenance Guide available at http://support.avaya.com

3.2. Disable RTP encryption

Execute the "list ip-codec-set" command.

list ip-codec-set

                                IP CODEC SETS

Codec  Codec 1     Codec 2     Codec 3     Codec 4     Codec 5
Set

  1    G.711MU
  2    G.711MU
  3    G.711MU
  4    G.711MU
  5    G.711MU
  6    G.711MU
  7    G.711MU

For each of codec sets execute the "change ip-codec-set N" command, where N is an index of set (from 1 to 7 in above example).

For example, to edit the first codec set, execute the "change ip-codec-set 1" command and make certain that Media Encryption list contains only a single value "none". If other values are presented there (for example, "aes"), then remove all other values except "none" as shown below:

change ip-codec-set 1                                           Page   1 of   2

                          IP Codec Set

    Codec Set: 1

    Audio        Silence      Frames   Packet
    Codec        Suppression  Per Pkt  Size(ms)
 1: G.711MU           n         2        20
 2:
 3:
 4:
 5:
 6:
 7:


     Media Encryption
 1: none
 2: 
 3:

Repeat this step for all the remaining codec sets.

3.3. Enable RTCP reporting

Enter the "change ip-network-region N" command, where N is an existing network region used for the agents' ip phones and softphones. Make certain that the RTCP Reporting Enabled field is set to "y", as shown below. The RTCP packets are used by MiaRec to map IP addresses to agent extensions.

change ip-network-region 1                                      Page   1 of  19
                               IP NETWORK REGION
  Region: 1
Location:         Authoritative Domain: voip.example.com
    Name:
MEDIA PARAMETERS                Intra-region IP-IP Direct Audio: yes
      Codec Set: 1              Inter-region IP-IP Direct Audio: yes
   UDP Port Min: 2048                      IP Audio Hairpinning? y
   UDP Port Max: 3029
DIFFSERV/TOS PARAMETERS                  RTCP Reporting Enabled? y
 Call Control PHB Value: 34      RTCP MONITOR SERVER PARAMETERS
        Audio PHB Value: 46       Use Default Server Parameters? y
        Video PHB Value: 26

3.4. Disable IP-IP Direct Audio (optional)

You can skip this step if recording of internal calls between IP phones is not needed.

Avaya Communication Manager supports "shuffling" of the media streams, which allows two IP phones to send media directly between each other bypassing the media gateway. "Shuffling" (IP-IP Direct Audio) should be disabled when internal calls between IP phones need to be recorded. IP-IP Direct Audio can be disabled either for individual IP phones or for a whole ip network region.

To disable IP-IP Direct Audio for individual IP phone, enter the "change station xxxxx", where xxxxx is phone's extension and change "Direct IP-IP Audio Connections" to "n" on Page 2.

Parameter "IP Audio Harpinning" is recommended to set to "y". In this case MedPro board acts as a proxy without allocating of resources on TDM bus.

change station 51001                                            Page   2 of   5
                                     STATION
FEATURE OPTIONS
           LWC Reception: spe              Auto Select Any Idle Appearance? n
          LWC Activation? y                         Coverage Msg Retrieval? y
  LWC Log External Calls? n                                    Auto Answer: none
             CDR Privacy? n                               Data Restriction? n
   Redirect Notification? y                     Idle Appearance Preference? n
 Per Button Ring Control? n                   Bridged Idle Line Preference? n
   Bridged Call Alerting? n                       Restrict Last Appearance? y
  Active Station Ringing: single
                                                         EMU Login Allowed? n
        H.320 Conversion? n          Per Station CPN - Send Calling Number?
       Service Link Mode: as-needed
         Multimedia Mode: enhanced
    MWI Served User Type:                       Display Client Redirection? n
              AUDIX Name:                      Select Last Used Appearance? n
             IP Hoteling? n                      Coverage After Forwarding? s
                                                   Multimedia Early Answer? n
 Remote Softphone Emergency Calls: as-on-local Direct IP-IP Audio Connections? n
  Emergency Location Ext: 51001         Always Use? n IP Audio Hairpinning? y

To disable IP-IP Direct Audio for all IP phones inside the IP network region, enter the "change ip-network-region N" command, where N is an existing network region used for the agents' ip phones and softphones and change "Intra-region IP-IP Direct Audio" and "Inter-region IP-IP Direct Audio" to "no".

Parameter "IP Audio Hairpinning" is recommended to set to "y".

change ip-network-region 1                                      Page   1 of  19
                               IP NETWORK REGION
  Region: 1
Location:         Authoritative Domain: voip.example.com
    Name:
MEDIA PARAMETERS                Intra-region IP-IP Direct Audio: no
      Codec Set: 1              Inter-region IP-IP Direct Audio: no
   UDP Port Min: 2048                      IP Audio Hairpinning? y
   UDP Port Max: 3029
DIFFSERV/TOS PARAMETERS                  RTCP Reporting Enabled? y
 Call Control PHB Value: 34      RTCP MONITOR SERVER PARAMETERS
        Audio PHB Value: 46       Use Default Server Parameters? y
        Video PHB Value: 26

Alternatively, instead of changing "IP-IP Direct Audio" parameter, you can change port mirroring configuration and mirror each IP phone's port rather than CLAN and MedPro ports (see below network diagram).

Such configuration of port mirroring allows a recording of internal calls without changing of "IP-IP Direct Audio" parameter.

3.5. Administer System Parameters Features

Enter the "change system-parameters features" command. Navigate to Page 5, and verify that Create Universal Call ID (UCID) has value “y”. If not, then set it to "y" and set UCID Network Node ID to an unassigned node ID.

change system-parameters features                               Page   5 of  17
                        FEATURE-RELATED SYSTEM PARAMETERS

SYSTEM PRINTER PARAMETERS
  Endpoint:               Lines Per Page: 60

SYSTEM-WIDE PARAMETERS
                                     Switch Name:
            Emergency Extension Forwarding (min): 10
          Enable Inter-Gateway Alternate Routing? n
Enable Dial Plan Transparency in Survivable Mode? n
                              COR to Use for DPT: station

MALICIOUS CALL TRACE PARAMETERS
               Apply MCT Warning Tone? n   MCT Voice Recorder Trunk Group:

SEND ALL CALLS OPTIONS
     Send All Calls Applies to: station    Auto Inspect on Send All Calls? n

UNIVERSAL CALL ID
     Create Universal Call ID (UCID)? y    UCID Network Node ID: 9999
     Copy UCID for Station Conference/Transfer? n

Navigate to Page 13, and set Send UCID to ASAI to "y". This parameter allows for the universal call ID to be sent to MiaRec call recording application.

change system-parameters features                               Page  13 of  17
                        FEATURE-RELATED SYSTEM PARAMETERS

 CALL CENTER MISCELLANEOUS
                         Clear Callr-info: next-call
        Allow Ringer-off with Auto-Answer? n

    Reporting for PC Non-Predictive Calls? n


  ASAI
            Copy ASAI UUI During Conference/Transfer? n
        Call Classification After Answer Supervision? n
                                   Send UCID to ASAI? y