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.

Station

This section contains the following topics:

Configuring Stations (J-Web Procedure)

To configure stations using the J-Web interface:

  1. Select Configure>Convergence Services>Station.

    The Station configuration page appears. Table 101 explains the contents of this page.

  2. Click one:
    • Add— Adds a new station configuration. Enter information as specified inTable 102.
    • Edit— Edits a selected station configuration. Enter information as specified inTable 102.
    • Delete— Deletes the selected station 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 101: Station Configuration page

Field

Function

Station name

Displays the station name.

Extension

Displays the extension name.

Template name

Displays the station template name.

Type

Displays the station type (Analog or SIP).

Class of restriction

Specifies the types of calls that can be made from the station; for example, local calls and long-distance calls. By default, intrabranch and emergency calls are always allowed.

TDM interface

Displays the time-division multiplexing interface.

Table 102: Add Station Configuration Details

Field

Function

Action

Name

Specifies the station name.

Enter the name of the station.

Extension

Specifies the station extension name.

Enter the name of the station extension.

Class of restriction

Defines the types of calls that can be made from the station; for example, local calls and long-distance calls.

Enter the class of restriction. By default, intrabranch and emergency calls are always allowed.

Caller iD

Specifies the caller ID information to be displayed to called parties.

Enter the caller ID details.

Direct inward dialing

Specifies the DID number that external users can dial to reach the extension directly, bypassing an operator, and information about the line to be used for DID.

Enter the DID number.

Authentication identifier

Specifies the authentication ID to authenticate itself to the station.

Enter an authentication ID.

Authentication password

Specifies the authentication password to authenticate itself to the station.

Enter an authentication password.

Template name

Specifies the name of the station template

Enter the name of the station template.

Enable analog

Enables the analog station.

Enable Analog check box.

Tdm interface

Defines the time-division multiplexing interface.

Enter the TDM interface.

T1 slot number

Specifies the slot connection in the channel bank.

Configuring Station Template (J-Web Procedure)

To configure templates using J-Web:

  1. Select Configure>Convergence Services >Station Template.

    The Station template configuration page appears. Table 103 explains the contents of this page.

  2. Click one:
    • Add—Adds a new station template configuration. Enter information as specified in Table 104.
    • Edit—Edits a selected station template configuration.
    • Delete—Deletes the selected station template 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 103: Station Template Configuration Page

Field

Function

Template name

Displays the station template name.

Type

Displays the type of station template.

Details of Station Template

Name

Displays the name of the station template selected.

Value

Displays the parameters selected in the station template.

Table 104: Add Station Template Configuration Details

Field

Function

Action

Template name

Specifies the station template name.

Enter the name of the station template.

Caller id transmit

Specifies the caller ID information configured for this parameter that is routed to all called parties.

Caller ID transmit is enabled by default. However, if it is disabled in a template that is applied to the station, caller ID information configured for the station is not transmitted.

Class of restriction

Specifies the types of calls that can be made from the station; for example, local calls and long-distance calls.

By default, intrabranch and emergency calls are always allowed.

Analog

Specifies the templates for analog stations.

Click for analog template.

SIP

Specifies the template for sip stations

Click for SIP template.

Analog

Voice activity detection

Specifies the number of packets sent when calls are idle.

Select the option ON or OFF

Comfort noise generation

Specifies the comfort noise generation transmitted instead of tones when a call is pending or placed on hold and when the parties are not speaking.

Select the option ON or OFF

SIP

Tone

Specifies the tone detection to enable calling features based on digit identification or tones generated by other analog devices or the PSTN.

The default tone is rfc-2833.

Codec

The following codec options are available:

  • G711-MU
  • G711-A
  • G729AB

Select one of the codec options.

Configuring Class of Restriction (J-Web Procedure)

To configure class of restriction using the J-Web interface:

  1. Select Configure>Convergence Services>Station.

    The Class of restriction configuration page appears. Table 105explains the contents of this page.

  2. Click one:
    • Add— Adds a new class of restriction configuration. Enter information as specified in Table 106.
    • Edit— Edits a selected class of restriction configuration.
    • Delete— Deletes the selected class of restriction 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 105: Class of Restriction Configuration Page

Field

Function

Class of restriction name

Displays the class of restriction name.

Permission

Displays the permission available for the class of restrictions.

Table 106: Add Class of Restriction Configuration Details

Field FunctionAction

Class of restriction

Specifies the class of restriction name.

Enter the class of restriction name.

Policy name

Displays the permissions available for the policy.

Permission

Displays the permissions available for the class of restrictions.

Add

Adds the new policy configuration.

Click the Add button to create a new class of restriction.

New policy configuration

Policy name

Specifies the name of the policy.

Enter the name of the policy.

Permission

Permissions available for the policy

Enter the permission details.

Calltype

Specifies the type of call. Call types available are:

  • Inter-branch-call
  • International-call
  • Local-call
  • Long-distance-call

Select one of the options.

Media Gateway

This section contains the following topics:

Configuring the Media Gateway (J-Web Procedure)

To configure the media gateway using the J-Web interface:

  1. Select Configure >Convergence Services >Media Gateway .

    The Media gateway configuration page appears. Table 107 explains the contents of this page.

  2. Click one of the following in the list pane:
    • Add — Adds a new media gateway configuration. Enter information as specified in Table 108.
    • Edit—Modifies the selected media gateway configuration.
    • Delete—Deletes the selected media gateway 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 107: Media Gateway Configuration Page

Field

Function

Media gateway name

Displays the media gateway name.

Description

Displays the details of the media gateway name.

Details of the Media Gateway

Name

Displays the name of the selected media gateway.

Value

Displays the details of the selected media gateway.

Table 108: Add New Media Gateway Configuration Details

Field

Function

Action

Media gateway

Specifies the media gateway name.

Enter the name of the media gateway.

Transport

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

Enter the transport value. The default transport is UDP.

Port number

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

Enter the port number. The default port is 5060.

Call Server

Specifies the peer call server name.

Enter the call server name.

Dialplan

Specifies the configuration for the routing of calls.

Enter the dial plan.

Zone

Specifies the service point for the services gateway.

Enter the zone.

Configuring Trunk Groups (J-Web Procedure)

To configure trunk groups using the J-Web interface:

  1. Select Configure>Convergence Services>Media Gateway>Trunk Groups.

    The Trunk group configuration page appears. Table 109 explains the contents of the page.

  2. Click one:
    • Add — Adds a new trunk group configuration. Enter information as specified in Table 110.
    • Edit — Edits the selected trunk group configuration.
    • Delete — Deletes the selected trunk group 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 109: Trunk Group Configuration Page

Field

Function

Trunk group name

Displays trunk group name.

Description

Displays a description of the trunk group.

Table 110: Add New Trunk Group Configuration Details

Field

Function

Action

Trunk group name

Specifies the trunk group name.

Enter the name of the trunk group.

Available trunks

Specifies the available trunk group.

Selected trunks

Specifies the selected trunk group.

Configuring Trunks (J-Web Procedure)

To configure trunks using the J-Web interface:

  1. Select Configure >Convergence Services >Media Gateway>Trunks.

    The Media gateway configuration page appears. Table 111 explains the contents of this page.

  2. Click one:
    • Add — Adds a new trunk configuration. Enter information as specified in Table 112.
    • Edit — Edits a selected trunk configuration.
    • Delete — Deletes the selected trunk configuration.
  3. Click one of the following buttons:
    • 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 111: Media Gateway Configuration Page

Field

Function

Trunk name

Displays the trunk name.

Trunk type

Displays the trunk type.

Details of trunk

Name

Displays the details of the trunk selected, such as trunk type, time slots, and signaling method.

Value

Displays actual values for the trunk details, such as fxo, 1-11, and t1.

Table 112: Add Trunk Gateway Configuration Details

Field

Function

Action

Trunk name

Specifies the trunk name.

Enter the name of the trunk.

Trunk type

Specifies the trunk type. The trunk types available are :

  • t1
  • fxo
  • fxs
  • t1
  • sip

Select one of the trunk types.

Fxo/Fxs/T1

TDM interface

Specifies the time-division multiplexing interface.

Enter the TDM interface.

Signaling type

Specifies the signaling type. The signaling types are:

  • fxo-loop-start
  • fxs-loop-start

Select the signaling option.

Time slots

Specifies the slot in the channel bank.

SIP

Codec

Specifies the codecs.

  • G711-MU
  • G711-A
  • G729AB

Select the codec options.

Sip

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.

Description

Specifies the trunk description

Enter a brief description.

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.

Port Number

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

Enter the port number. The default port is 5060.

Transport

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

Enter the transport. The default used is UDP

Dial Plan

This section contains the following topics:

Configuring a Dial Plan (J-Web Procedure)

To configure a dial plan using the J-Web interface:

  1. Select Configure>Convergence Services>Dial Plan.

    The Dialplan configuration page appears. Table 113 explains the contents of this page.

  2. Click one:
    • Add — Adds a new dial plan configuration. Enter information as specified in Table 114.
    • Edit—Edits the selected dial plan configuration.
    • Delete—Deletes the selected dial plan 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 113: Dial Plan Configuration Page

Field

Function

Dialplan name

Displays the dial plan name.

Preferences

Displays the preferences selected.

Digit Transformation

Displays the digital transform selected.

Table 114: Add Dial Plan Configuration Details

Field

Function

Action

Dialplan name

Specifies the dial plan name.

Enter the name of the dial plan.

Route pattern name

Specifies the route pattern name.

Enter the name of the route pattern.

Add>New Route Pattern Configuration

Route pattern

Specifies the route of the call and offers multiple paths if multiple trunks are configured for the route pattern.

Enter the route pattern.

Call type

Specifies the type of call. Call types available are:

  • Inter-branch-call
  • International call
  • Local-call
  • Long-distance-call

Select one of the options.

Trunk Group Information

Preference

Specifies the call routing option.

Enter the preference. The preference is 0 by default.

Digit manipulation list

The options available are:

  • dgttrm
  • dgt1
  • dgt2
  • dgt7

Select one of the options.

Available trunk groups

Displays the available trunk group.

Trunk group

Displays the group of trunks to be used to route calls.

Preference

Displays preferences of the selected trunk group.

Digit transform

Displays the digital transform of the selected trunk group.

Configuring Digital Manipulation (J-Web Procedure)

To configure digital manipulation using the J-Web interface:

  1. Select Configure>Convergence Services>Dial Plan>Manipulation.

    The digital manipulation configuration page appears. Table 115 explains the contents of this page.

  2. Click one:
    • Add — Adds a new digital manipulation configuration. Enter information as specified in Table 116.
    • Edit — Edits the selected digital manipulation configuration. Enter information as specified in Table 116.
    • Delete — Deletes the selected digital manipulation 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 115: Digital Manipulation Configuration page

Field

Function

Digit transform name

Displays the digit transformation name.

Regular expression

Displays the regular expression.

Table 116: Add Digital Manipulation Configuration Details

Field

Function

Action

Digit transformation name

Specifies the name of the digital manipulation configuration to be used by the dial plan for routing calls.

Enter the digit transform name.

Regular expression

Specifies the digit to be masked while routing calls.

Enter the regular expression.

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.