Convergence Series

This section contains the following topics:

Configuring Peer Call Servers (J-Web Procedure)

To configure peer call servers using the J-Web interface:

  1. Select Configure>Convergence Services>Call Service.

    The Peer call server configuration page appears. If you select a specific peer call server configuration, the peer call server details are displayed. Table 99 explains the contents of this page.

  2. Click one:
    • Add—Adds a new peer call server configuration. Enter information as specified in Table 100.
    • Edit—Edits a selected call server configuration.
    • Delete—Deletes the selected call server configuration.
  3. Click one:
    • OK—Saves the configuration and returns to the main configuration page.
    • Cancel—Cancels your entries and returns to the main configuration page.
    • Commit— Applies the configuration and other pending changes (if any).

Table 99: Peer Call Server Configuration Page

Field

Function

Callserver

Displays the call server name.

Protocol

Displays the protocol.

Description

Displays a description of the protocol.

Details of Call Server

Name

Displays the details of the call server selected.

Value

Displays the actual values of the call server selected.

Table 100: Add New Peer Call Server Configuration Details

Field

Function

Action

Name

Specifies the new call server name.

Enter the name of the call server.

Description

Describes the new call server.

Enter the description of the call server.

Port number

Specifies the ports to be associated with this VLAN for voice traffic.

Enter the port number. The default port number is 5060.

Transport

Specifies the transport used for the SIP protocol: UDP, TCP, or TLS.

Enter the transport. The default transport is UDP.

Tone generation

Specifies the tones generated by other analog devices or the PSTN.

Enter the tone generation value. The default option used for tone generation is rfc-2833.

SIP

Codec

Specifies the codec option to be selected for the peer call server configuration. The options are:

  • G711-MU
  • G711-A
  • G729AB

Select one of the options.

Authentication identifier

Specifies the authentication ID to authenticate itself to the peer call server.

Enter an authentication ID.

Authentication password

Specifies the authentication password to authenticate itself to the peer call server.

Enter an authentication password.

PSTN access number

Specifies the access codes to identify calls to be routed out PSTN trunks.

Enter the PSTN access number.

Address

Address type

Specifies the peer call server address. The options available are:

  • Fqdn
  • ipv4-sddr

Select the address type required.

Fqdn

Specifies the fully qualified domain name.

Enter the FQDN. You can use an FQDN when multiple peer call servers are to be used.

IP address

Specifies the IPv4 address.

Enter the IPv4 address.

Configuring Call Features (J-Web Procedure)

To configure call features using the J-Web interface:

  1. Select Configure>Convergence Services>Call Features.

    The Call features configuration page appears. Table 117explains the contents of this page.

  2. Click one:
    • Edit— Modifies the selected call features configuration. Enter information as specified in Table 118.
    • Delete— Deletes the selected call features configuration.
  3. Click one
    • OK—Save the configuration and returns to the main configuration page.
    • Cancel—Cancel your entries and returns to the main configuration page.
    • Commit— Applies the configuration and other pending changes (if any).

Table 117: Call Features Configuration Page

Field

Function

Name

Displays the feature selected.

Value

Displays the value of the call configuration.

Table 118: Edit Options

Field

Function

Action

Live Attendant

Extension

Specifies the extension number to which the call must be placed.

Enter the extension number.

Date From

Specifies the start date.

Enter the from date.

Date To

Specifies the end date.

Enter the to date.

Auto Attendant

Ring Count

Specify the number of times the telephone for the person referred to as the live attendant may ring before the call is forwarded to auto-attendant for handling.

Enter the number of ring.

Voicemail

Extension

Specify the extension number for users to call to retrieve their voicemail when the survivable call server is in control.

Enter the extension number of the voicemail.

Remote Access Number

Specify the PSTN remote access number of the voicemail server (when peer call server provides voicemail services)

Enter the remote access number of peer call server.

Guest

Auto Register

  

Class of Restriction

  
Music on hold

Directory

  

Format

  

Order

  

Configuring Survivable Call Service (J-Web Procedure)

To configure survivable call service using the J-Web interface:

  1. Select Configure>Convergence Services>Call Server.

    The Survivable callservice configuration page appears. Table 119 explains the contents of this page.

  2. Click one:
    • Add— Adds a new call service configuration. Enter information as specified in Table 120.
    • Edit—Edits the selected call service configuration.
    • Delete—Deletes the selected call service configuration.
  3. Click one:
    • OK—Saves the configuration and returns to the main configuration page.
    • Cancel—Cancels your entries and returns to the main configuration page.
    • Commit—Applies the configuration and other pending changes (if any).

Table 119: Survivable Call Service Configuration Page

Field

Function

Callservice name

Displays the survivable call service name.

Peer callserver name

Displays the peer call server configuration name.

Details of the survivable callservice

Name

Displays the details for the survivable call service selected.

Value

Displays the actual values of the configuration selected.

Table 120: Add New Call Service Configuration Details

Field

Function

Action

Callserver name

Specifies the call service name.

Enter the name for the call service.

Transport

Specifies the transport used for the SIP protocol: UDP or TCP.

Enter the transport details. The default used is UDP.

Port number

Specifies the number used for the port or communications endpoint for the SIP protocol.

Enter the port number. The default port number is 5060.

Registration expiry timeout

Specifies the time duration required for the SRX Series SCS to accept registrations from SIP stations and redirect any call requests to the peer call server after the peer call server has regained control.

Enter a number from 30 to 86,400seconds. The default is 60 seconds.

Heartbeat normal

Determines the interval when the SRX Series SCS sends keepalive messages to the peer call server. The heartbeat parameter works in conjunction with the SIP timeout parameter.

Enter a number from 2 to 8 seconds. The default is 2 seconds.

Heartbeat survivable

Specifies the time when the SRX Series SCS sends keepalive messages to the peer call server to determine if it is reachable and has recovered from the fault condition. After the peer call server responds, the SRX Series SCS enters a watch period to determine if it is reliably reachable.

Enter a number from 800 to 4000milliseconds. The default is 800 milliseconds.

Response threshold

Specifies the minimum percent of time the peer call server must respond to timeout messages during the watch period.

Enter a number from 10 to 100 percent. The default is 75 percent.

Callserver

Specifies the peer call server.

Enter one of the options: fully qualified domain name (FQDN) or IPv4 address.

Dialplan

Specifies the configuration for how calls are to be routed.

Enter the dial plan.

Service point

Zone

Specifies the service point to enable Integrated Convergence Services for the security zone.

Enter the service point details.