Configuring FXS Ports for Basic Calls

First Published: October 2, 2008
Last updated: September 4, 2015

 

This module describes how to configure analog Foreign Exchange Station (FXS) ports on a Cisco Integrated Services Router (ISR) or Cisco VG224 Analog Phone Gateway for basic calls.

Finding Feature Information in This Module

Your Cisco IOS software release may not support all of the features documented in this module. To reach links to specific feature documentation and to see a list of the releases in which each feature is supported, see the “Feature Information for Configuring FXS Ports for Basic Calls” section.

Finding Support Information for Platforms and Cisco IOS Software Images

Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Contents

Prerequisites for Configuring FXS Ports for Basic Calls

Cisco IOS Gateway

Analog Endpoints in Cisco Unified Communications Manager

  • Cisco Unified Communications Manager 4.2 or a later version.
  • Cisco voice gateway analog Foreign Exchange Station (FXS) ports are added in Cisco Unified Communications Manager. Each analog FXS port on which SCCP is enabled counts as a single IP phone for licensing purposes. For example, to register all 24 ports on a Cisco VG224 Analog Phone Gateway, the 24 ports count toward the 2500 limit if you purchase a Cisco Unified Communications Manager license for 2500 devices.

Analog Endpoints in Cisco Unified CME

  • Basic feature license and phone user licenses are purchased. For licensing purposes, each analog FXS port on which SCCP is enabled counts as a single phone.

Information About FXS Ports for Basic Calls

To configure FXS ports for basic calls, you should understand the following concepts:

Hookflash Duration

Analog phones use hookflash to access a second dial tone to initiate certain SCCP phone features such as transfer and conference. Hookflash is an on-hook condition of short duration that is usually generated when a phone user presses the Flash button on a phone. The duration of an on-hook condition generated by a Flash button varies for different phone makes and models. Cisco voice gateways measure the duration of detected on-hook conditions to determine whether they should be interpreted as hookflash or not. The duration of a detected on-hook condition is interpreted by Cisco IOS software as follows:

  • An on-hook condition that lasts for a time period that falls inside the hookflash duration range is considered a hookflash.
  • An on-hook condition that lasts for a shorter period than the lower limit of the range is ignored.
  • An on-hook condition that lasts for a longer period than the higher limit of the range is considered a disconnect.

The hookflash duration range for FXS ports is defined as follows:

  • The lower limit of the range is set in software at 150 ms, although there is also a hardware-imposed lower limit that is typically about 20 ms, depending on platform type. An on-hook condition that lasts for a shorter time than this hardware-imposed lower limit is not reported to the Cisco IOS software.
  • The upper limit of the range is set in software at 1000 ms by default, although this value can be changed on the voice gateway. The upper limit can be set to any value from 50 to 1550 ms.
  • If the upper limit of the hookflash duration range is X, a value greater than 150, then any on-hook duration between 150 and X is interpreted as a hookflash. For example, if X is 1550, the hookflash duration range is 150 to 1550 ms. An on-hook signal that lasts for 1250 ms is interpreted as a hookflash, and an on-hook signal of 55 ms is ignored.
  • If the upper limit of the hookflash duration range is X, a value less than 150, then any on-hook duration between Y, the hardware lower limit, and X is interpreted as a hookflash. For example, if X is 65, the hookflash duration range is Y to 65 ms (assume Y is 20 ms). An on-hook signal that lasts for 1250 ms is interpreted as a disconnect, and an on-hook signal of 55 ms is interpreted as a hookflash. An on-hook signal of less than Y is ignored.

For information about modifying the upper limit of the hookflash duration range, see the “Modifying Hookflash” section.

PLAR with DTMF Out-Pulse Digits

A private line automatic ring-down (PLAR) connection allows an analog phone user to make a call without dialing any digits. When the user goes off-hook on the phone, the Cisco voice gateway automatically rings a predefined extension or PSTN number. The PLAR number is configured on the analog FXS port to which the corresponding analog phone is connected.

The PLAR with DTMF out-pulse digits feature in Cisco IOS Release 12.4(9)T is an enhancement that enables the voice gateway to out-pulse additional DTMF digits after the PLAR connection is up. These DTMF digits are configurable and can include 0 to 9, A to D, a comma (,) for a one-second pause, an asterisk (*), and number sign (#). If an analog phone user presses a string of digits (0-9, *, #) after taking a PLAR phone off-hook, the voice gateway buffers the digit string until the DTMF digits are done being out-pulsed. After the voice gateway sends all the DTMF digits, it sends the buffered digits to the destination port.

Although users do not hear a dial tone when taking a PLAR phone off-hook, PLAR phones support the same features as other analog phones. PLAR phones can receive incoming calls and support hookflash for basic supplementary features such as call transfer, call waiting, and conference. Feature access codes (FACs) and speed-dial codes are not valid immediately after taking a PLAR phone off-hook, but after connecting to the destination port, a user can press hookflash to get a dial tone and then dial an access code for features such as speed dial, redial, and call transfer.

For configuration information, see the “Configuring PLAR with DTMF Out-Pulse Digits” section.

Dial Tone Generation after Remote Onhook

The Dial Tone Generation after Remote Onhook feature provides PBX interoperability by enabling configurable automatic dial tone capability after remote call disconnect. Dial tone is automatically generated to the remaining party in a basic A-B call scenario once one party disconnects, in the same way that a PBX user gets immediate dial tone after remote party disconnect. This allows the user to make a new call without hookflash or going onhook, then off hook. If automatic dial tone generation is disabled, the user is required to go onhook then off-hook, or perform a hookflash, in order to make a new call.

After remote onhook there are two ways for an SCCP analog phone to redial, either by the user pressing a redial button or entering a feature access code (FAC). Some phone model redial buttons do not function with the dial tone after remote onhook feature enabled, resulting in redial digits not being sent. For this reason, the dial tone generation after remote onhook feature supports redial only when activated by FAC.

Dial tone generation immediately after the remote party goes onhook is configurable on a per port basis and is enabled by default. Automatic dial tone is supported only on STC application-controlled loop start FXS ports. For devices such as interactive voice response (IVR) systems that require power denial to disconnect properly, power denial is triggered prior to dial tone generation after one party disconnects. For a PLAR port, dial tone is played instead of triggering another PLAR after remote party disconnect. You cannot configure the dial tone generation after remote onhook feature using the Cisco Unified Communications Manager auto configuration capability.

For configuration information, see the “Configuring SCCP Gateway Dial Tone Generation After Remote Onhook” section.

Ground Start FXS Ports as SCCP Analog Endpoints

SCCP enhanced supplementary features provide support on the SCCP analog gateway for ground start FXS ports, used for PBX and key system connections, enabling disconnect supervision and Cisco Unified Communications Manager registration. The ground start FXS port feature is supported for basic calls only and supports PBX interoperability by providing supervisory disconnect to FXS ports and analog endpoints. Prior to ground start FXS support, there was no disconnect supervision to signal the end of a call. The ground start FXS ports feature provides power denial-based supervisory disconnect to indicate remote party disconnect using the loop current feed open (LCFO) mechanism.

For configuration information, see the “Configuring SCCP Gateway Ground Start FXS Ports” section.

Supervisory Disconnect

The Supervisory Disconnect feature provides a disconnect indication to the remote party in a two-party call after one side disconnects. This enables external applications connected to the Cisco voice gateway to promptly clear a call after receiving the disconnect indication. This feature triggers a power denial on FXS ports with loop-start signaling when a voice call disconnects. The power denial is only generated in a two-party call scenario when one party disconnects. Power denial is not generated if a call is on hold and either the active party or the on-hold party hangs up. Power denial is also not generated for a three-way conference call when one party hangs up. This feature is enabled and disabled on a voice port basis. The remote party receives the power denial signal for the duration that is set on the analog FXS port.

Because the Cisco voice gateway cannot distinguish the type of device connected to the analog FXS port, a power denial signal is sent to all FXS ports that have the power denial feature enabled. This can result in analog phones also receiving a power denial signal after one party disconnects in a two-party call. The remaining party hears a brief click sound. To prevent this behavior on analog phones, you can disable the power denial feature on the analog FSX voice port. For configuration information, see the “Configuring Supervisory Disconnect” section.

How to Configure FXS Ports for Basic Calls

note.gif

Noteblank.gif This document does not contain details about configuring Cisco Unified Communications Manager or Cisco Unified CME. See the documentation for these products for installation and configuration instructions.

This section contains the following tasks for setting up SCCP analog phone support:

Enabling SCCP on the Voice Gateway

To enable SCCP on the local interface that communicates with your Cisco call-control system and identify priority levels to Cisco Unified Communications Manager servers or Cisco Unified CME routers, perform the following steps.

note.gif

Noteblank.gif If more than 72 end points are configured with SCCP in a single voice gateway, we recommend you to increase the hold-queue size on the interface of the gateway to 300.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif sccp local interface-type interface-number [ port port-number ]

4.blank.gif sccp ccm { ip-address | dns } identifier identifier-number [ port port-number ] [ version version-number ]

5.blank.gif sccp

6.blank.gif sccp ccm group group-number

7.blank.gif associate ccm identifier-number priority priority-number

8.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

sccp local interface-type interface-number [ port port-number ]

 

Router(config)# sccp local FastEthernet0/0

Selects the local interface that SCCP applications (transcoding and conference) use to register with Cisco Unified Communications Manager and Cisco Unified CME.

  • port-number —(Optional) TCP or UDP port number used by the selected interface. Range: 1025 to 65535. Default: 2000.

Step 4

sccp ccm { ip-address | dns } identifier identifier-number [ port port-number ] [ version version-number ]

 

Router(config)# sccp ccm 10.8.1.2 identifier 10 version 4.1

Adds a Cisco Unified Communications Manager server or Cisco Unified CME router to the list of available call-control systems.

  • Repeat this step to add a backup system.

Step 5

sccp

 

Router(config)# sccp

Enables SCCP and its related applications (transcoding and conferencing).

Step 6

sccp ccm group group-number

 

Router(config)# sccp ccm group 1

Creates a group of Cisco Unified Communications Manager or Cisco Unified CME systems and enters SCCP ccm configuration mode.

  • group-number —Number that identifies the group. Range: 1 to 50.
  • Repeat this step to add a backup system.

Step 7

associate ccm identifier-number priority priority-number

 

Router(config-sccp-ccm)# associate ccm 1 priority 1

Adds a Cisco Unified Communications Manager server or Cisco Unified CME router to the group and establishes its priority within the group.

  • identifier-number —Identifier that was defined in the sccp ccm command in Step 4.
  • priority priority-number —Number that indicates the priority of this Cisco Unified Communications Manager server or Cisco Unified CME router. Range: 1 to 4, where 1 is the highest priority.

Note A second Cisco Unified Communications Manager or Cisco Unified CME with a lower priority number becomes a backup system.

Note The priority must match the order of the call manager group associated with this device within call manager.

Step 8

registration timeout timeout-value

 

Router(config-sccp-ccm)# registration timeout 3

(Optional) Sets the length of time between registration messages sent from SCCP to the Cisco Unified Communications Manager.

  • Time, in seconds, between registration messages. Range is 1 to 180. Default is 3.

Step 9

keepalive timeout seconds

 

Router(config-sccp-ccm)# keepalive timeout 3

(Optional) Sets the length of time between keepalive messages from Skinny Client Control Protocol (SCCP) to Cisco Unified Communications Manager.

  • Time between keepalive messages. Range is 1 to 180. Default is 30.

Step 10

connect interval seconds

 

Router(config-sccp-ccm)# connect interval 3

(Optional) Specifies the amount of time that a given digital signal processor (DSP) farm profile waits before attempting to connect to a Cisco Unified Communications Manager when the current Cisco Unified Communications Manager fails to connect.

  • Timer value, in seconds. Range is 1 to 3600. Default is 60.

Step 11

switchback method graceful

 

Router(config-sccp-ccm)# switchback method graceful

(Optional) Sets the Cisco Unified Communications Manager switchback method.

  • graceful —The Cisco Unified Communications Manager switchback happens only after all the active sessions are terminated gracefully.

Step 12

end

 

Router(config-sccp-ccm)# end

Exits SCCP ccm configuration mode and returns to privileged EXEC mode.

Examples

The following example shows the configuration for SCCP communication on Cisco VG224 Fast Ethernet interface 0/0 to two Cisco Unified Communications Manager servers.

Router# show running-config
.
.
.
sccp local FastEthernet0/0
sccp ccm 10.4.13.20 identifier 10
sccp ccm 10.4.13.70 identifier 12
sccp
!
sccp ccm group 1
associate ccm 10 priority 1
associate ccm 12 priority 2
!

Enabling the STC Application for Analog FXS Ports

To enable the SCCP telephony control (STC) application and configure analog voice ports on the voice gateway for control by the STC application, perform the following steps.

Prerequisites

  • SCCP is enabled on the Cisco voice gateway. For configuration information, see the “Enabling SCCP on the Voice Gateway” section.
  • To enable the STC application for analog ports on a voice gateway on which the station-id number command is configured, remove the configuration for the station-id number command before performing this task.
note.gif

Noteblank.gif Only for FXS voice ports on a Cisco VG224 that is used with Cisco Unified CME or for FXS voice ports that are on a different router from Cisco Unified CME: To retain the station-id number configuration on your voice gateway, configure the answer-address command and do not remove the configuration for the station-id number command before performing this task.

Restrictions

  • If Cisco Unified CME and the FXS voice ports to be controlled by the STC application are on the same voice gateway and the station-id number and destination-pattern commands are already configured on that gateway, the dial-peer matches the wrong entry and cannot access the STC application. To enable the STC application for FXS ports on a voice gateway on which Cisco Unified CME is configured, remove the configuration for the station-id number command before performing this task.
  • If Cisco Unified CME and the FXS voice ports to be controlled by the STC application are on the same voice gateway and the station-id number command is configured for a voice-port which is controlled by the STC application with FACs, any feature code or speed dial code with ** will drop the call immediately. To use FACs that include **, remove the configuration for the station-id number command before performing this task.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif stcapp ccm-group group-number

4.blank.gif stcapp

5.blank.gif dial-peer voice tag pots

6.blank.gif service stcapp

7.blank.gif port slot-number / port-number

8.blank.gif exit

9.blank.gif voice-port slot-number/port-number

10.blank.gif caller-id enable

11.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

stcapp ccm-group group-id

 

Router(config)# stcapp ccm-group 1

Associates the STC application with a specific Cisco Unified Communications Manager group that controls calls and features.

Step 4

stcapp

 

Router(config)# stcapp

Enables the STC application.

Step 5

dial-peer voice tag pots

 

Router(config)# dial-peer voice 102 pots

Defines a specific dial peer and enters dial-peer configuration mode.

  • tag —Number that identifies the dial peer. Range: 1 to 2147483647.

Step 6

service stcapp

 

Router(config-dial-peer)# service stcapp

Enables the STC application on the dial peer.

Step 7

port slot-number / port-number

or

port slot-number / subunit-number/port

 

Router(config-dial-peer)# port 2/2

or

Router(config-dial-peer)# port 0/1/0

Assigns an analog voice port to the dial peer.

  • Format and values for analog FXS voice port number is platform-dependent. Type ? for values.

Step 8

exit

 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 9

voice-port slot-number / port-number

 

Router(config)# voice-port 2/0

Enters voice-port configuration mode.

  • slot-number / port-number —Analog FXS voice port number. Range: 2/0 to 2/23.

Step 10

caller-id enable

 

Router(config-voiceport)# caller-id enable

(Optional) Enables caller ID for this voice port.

Note Other parameters, such as caller-ID name and number, must be configured on the Cisco call-control system.

Step 11

end

 

Router(config-voiceport)# end

Exits voice-port configuration mode and returns to privileged EXEC mode.

Examples

The following example enables the STC application for Cisco Unified Communications Manager group 1 and associates the STC application with dial peer 102, to which the Cisco VG224 analog FXS port 2/2 has been assigned. This configuration also enables caller ID on voice port 2/2.

Router# show running-config
.
.
.
stcapp ccm-group 1
stcapp
!
dial-peer voice 102 pots
service stcapp
port 2/2
!
voice-port 2/2
caller-id enable
.
.
.

Modifying Hookflash

To change the upper limit of the hookflash duration range for an analog FXS port, perform the following steps.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif voice-port port-number

4.blank.gif timing hookflash-input milliseconds

5.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice-port slot-number / port-number

or

voice-port slot-number / subunit-number / port

 

Router(config)# voice-port 2/1

Enters voice-port configuration mode.

  • Format and values for analog FXS voice port number is platform-dependent. Type ? for values.

Step 4

timing hookflash-input milliseconds

 

Router(config-voiceport)# timing hookflash-input 175

Specifies the maximum duration of an on-hook condition that will be interpreted as a hookflash.

  • milliseconds —Range: 50 to 1550. Default: 1000.

Step 5

end

 

Router(config-voiceport)# end

Exits voice-port configuration mode and returns to privileged EXEC mode.

Configuring PLAR with DTMF Out-Pulse Digits

To configure an analog foreign exchange station (FXS) port to support PLAR, perform the following steps.

Prerequisites

  • Cisco IOS Release 12.4(6)T or a later release for Cisco VG224 Analog Phone Gateways.
  • Cisco IOS Release 12.4(9)T or a later release for Cisco ISRs.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif dial-peer voice tag pots

4.blank.gif service stcapp

5.blank.gif port slot-number / port

6.blank.gif exit

7.blank.gif sccp plar

8.blank.gif voiceport port-number dial dial-string [ digit dtmf-digits [ wait-connect wait-msecs ] [ interval inter-digit-msecs ]]

9.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

dial-peer voice tag pots

 

Router(config)# dial-peer voice 102 pots

Defines a specific dial peer and enters dial-peer configuration mode.

  • tag —Number that identifies the dial peer. Range: 1 to 2147483647.

Step 4

service stcapp

 

Router(config-dial-peer)# service stcapp

Enables the STC application for the dial peer.

Step 5

port slot-number / port

 

Router(config-dial-peer)# port 2/2

Assigns a voice port to the dial peer.

Step 6

exit

 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 7

sccp plar

 

Router(config)# sccp plar

Enters SCCP PLAR configuration mode.

Step 8

voiceport port-number dial dial-string [ digit dtmf-digits [ wait-connect wait-msecs ] [ interval inter-digit-msecs ]]

 

Router(config-sccp-plar)# voiceport 2/1 dial 3660 digit 1234 wait-connect 100 interval 100

Enables PLAR on analog FXS ports that use SCCP for call control.

  • port-number —Analog FXS voice port number. Range is 2/0 to 2/23.
  • dial dial-string —Up to 16 characters that can be dialed on a telephone keypad (0 to 9, A to D, *, #). The voice gateway sends this string to the call-control system when the analog phone goes off hook.
  • digit dtmf-digits —Up to 16 characters (0 to 9, A to D, *, #, and a comma (,) for a one-second pause). The voice gateway sends this string to the call-control system after the wait-msecs expires.
  • wait-connect wait-msecs —Number of milliseconds that the voice gateway waits after voice cut-through before out-pulsing the DTMF digits. Range: 0 to 30000, in multiples of 50. Default: 50. If 0, DTMF digits are sent automatically by the voice gateway after the call is connected.
  • interval inter-digit-msecs —Number of milliseconds between the DTMF digits. Range: 50 to 500, in multiples of 50. Default: 50.

Step 9

end

 

Router(config-sccp-plar)# end

Exits SCCP PLAR configuration mode and returns to privileged EXEC mode.

Examples

The following example shows PLAR enabled on voice port 2/0 and 2/1.

Router# show running-config
.
.
.
sccp plar
voiceport 2/0 dial 3660 digit 1234 wait-connect 100 interval 100
voiceport 2/1 dial 3660 digit 6789 interval 100
!
!
!
dial-peer voice 500 pots
service stcapp
port 2/0
!
dial-peer voice 501 pots
service stcapp
port 2/1
!

Configuring SCCP Gateway Dial Tone Generation After Remote Onhook

This task configures dial tone generation after remote onhook. This feature enables the SCCP gateway to generate dial tone to the remaining party in basic call mode once the remote party disconnects. Perform this task to allow PBX interoperability by enabling configurable automatic dial tone capability after remote call disconnect.

Prerequisites

  • Cisco IOS Release 12.(6)XE or a later release.

Restrictions

  • The SCCP Gateway Dial Tone Generation After Remote Onhook feature is supported only on SCCP loop-start FXS ports that are registered with Cisco Unified Communications Manager and Cisco Unified CME.
  • You cannot configure the SCCP Gateway Dial Tone Generation After Remote Onhook feature using Cisco Unified Communications Manager automatic download capability.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif dial-peer voice tag pots

4.blank.gif tone dialtone remote-onhook

5.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

dial-peer voice tag pots

 

Router(config)# dial-peer voice 101 pots

Defines a particular dial peer and enters dial peer voice configuration mode.

  • tag —Number that identifies the dial peer range: 1 to 2147483647.

Step 4

tone dialtone remote-onhook

 

Router(config-dial-peer)# tone dialtone remote-onhook

Enables dial tone generation after remote onhook.

Step 5

end

 

Router(config-dial-peer)# end

Exits dial-peer configuration mode and returns to privileged EXEC mode.

Examples

The following examples show the dial tone generation after the remote onhook feature is enabled. Because the dial tone generation after remote onhook feature is enabled by default, it does not display in the show running-config output.

 
Router# show running-config
 
service stcapp
dial-peer voice 3001 pots
port 1/1/1
 
Router# show dial-peer voice 3001
 
VoiceEncapPeer3001
peer type = voice, system default peer = FALSE, information type = voice,
!
!
!
in bound application associated: 'stcapp'
dial tone generation after remote-onhook = enabled
 
Router# show stcapp device voice-port 1/1/1
 
Port Identifier: 1/1/1
!
Dialtone after remote-onhook feature: activated

 

The following examples show the dial tone generation after the remote onhook feature disabled.

Router# show running-config
 
no tone dialtone remote-onhook
dial-peer voice 3002 pots
service stcapp
port 1/1/0
 
Router# show dial-peer voice 3002
 
VoiceEncapPeer3002
!
dial tone generation after remote-onhook = disabled
 
Router# show stcapp device voice-port 1/1/0
 
Port Identifier: 1/1/0
!
Dialtone after remote-onhook feature: not activated

Troubleshooting Tips

The following commands can help troubleshoot the dial tone after remote onhook feature:

  • debug voip application stcapp all
  • debug voip application stcapp port port-number

Configuring SCCP Gateway Ground Start FXS Ports

To configure ground-start FXS ports, perform the following steps.

Prerequisites

  • Cisco IOS Release 12.4(6)XE or a later release.

Restrictions

  • Supports for basic calls only.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif voice-port [ slot-number/ subunit-number/port ]

4.blank.gif signal ground-start

5.blank.gif shutdown

6.blank.gif no shutdown

7.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice-port [slot-number/subunit-number/port]

 

Router(config)# voice-port 1/1/1

Enters voice-port configuration mode.

Step 4

signal ground-start

 

Router(config-voiceport)# signal ground-start

Configures ground-start signaling as the signaling type for the voice port.

Step 5

shutdown

 

Router(config-voiceport)# shutdown

Takes the specified voice port offline and triggers deregistration of the device with Cisco Unified Communications Manager.

Step 6

no shutdown

 

Router(config-voiceport)# no shutdown

Puts the specified voice port back online and triggers reregistration of the device with Cisco Unified Communications Manager.

Step 7

end

 

Router(config-voiceport)# end

Exits voice-port configuration mode and returns to privileged EXEC mode.

Examples

The following example shows the ground start FXS port feature enabled and verifies the port type:

Router# show voice port 1/1/1
 
Foreign Exchange Station 1/1/1 Slot is 1, Sub-unit is 1, Port is 1
Type of VoicePort is FXS VIC2-2FXS
Operation State is DORMANT
Administrative State is UP
!
Voice card specific Info Follows:
Signal Type is groundStart

Troubleshooting Tips Ground Start FXS Ports

The following command can help troubleshoot ground start FXS ports:

  • debug vpm signal—Use this command to collect debugging information for signaling events.

Configuring Supervisory Disconnect

To configure the Supervisory Disconnect feature on an analog FXS voice port, perform the following steps.

SUMMARY STEPS

1.blank.gif enable

2.blank.gif configure terminal

3.blank.gif voice-port port-number

4.blank.gif supervisory disconnect lcfo

5.blank.gif timeouts power-denial ms

6.blank.gif end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice-port port-number

 

Router(config)# voiceport 2/1

Enters voice port configuration mode.

  • port-number —Format and values for analog FXS voice port number is platform-dependent. Type ? for values.

Step 4

supervisory disconnect lcfo

 

Router(config-voiceport)# supervisory disconnect lcfo

Sends a loop current feed open (LCFO) signal to indicate a disconnect on an FXS loop-start port.

Note This command is enabled by default.

Step 5

timeouts power-denial ms

 

Router(config-voiceport)# timeouts power-denial 500

Sets the duration of the power denial timeout on the specified FXS voice port.

  • ms —Number of milliseconds that the power-denial is invoked. Range: 0 to 2500. Default: 750.

Step 6

end

 

Router(config-voiceport)# end

Exits voice-port configuration mode and returns to privileged EXEC mode.

Examples

The following example shows that the duration of the power denial is 500 ms on port 2/0 (which has supervisory disconnect enabled by default) and supervisory disconnect on port 2/1 is disabled.

Router# show running-config
.
.
.
voice-port 2/0
timeouts power-denial 500
!
voice-port 2/1
no supervisory disconnect
.
.
.

Verifying and Troubleshooting the Configuration

Use the following commands on the voice gateway to verify the configuration and status of the STC application and SCCP:

  • show call application voice summary —Displays whether the STC application is running.
  • show call application voice stcapp —Displays detailed application state and statistics.
  • show call active voice —Displays the number of calls that are currently active. Call legs associated with this feature are included in the “Call agent controlled call-legs” listing.
  • show sccp [ all | connections | statistics ]—Displays SCCP information such as administrative and operational status.
  • show stcapp device summary —Displays a summary of endpoints associated with the STC application, and their states, types, and directory numbers.
  • show stcapp device [ name device-name | voice-port port ]—Displays information for a single endpoint associated with the STC application. If an active call is in progress, the output will display additional call-related information.
  • show stcapp statistics [ all | voice-port port ]—Displays call statistics for endpoints associated with the STC application.
  • show running-config —Displays running configuration nondefault values.

Use the following commands on the voice gateway to troubleshoot the STC application and SCCP:

  • debug [ voip | voice ] application stcapp all —Displays detailed debugging for all ports.
  • debug [ voip | voice ] application stcapp error —Displays error debugging for all ports.
  • debug [ voip | voice ] application stcapp events —Displays call flow event debugging for all ports.
  • debug [ voip | voice ] application stcapp functions —Displays function debugging for all ports.
  • debug [ voip | voice ] application stcapp port port —Displays detailed debugging only for the specified port.
  • debug sccp all —Displays detailed debugging for all SCCP debug trace information.
  • debug sccp config —Displays SCCP auto-configuration/download debugging.
  • debug sccp errors —Displays SCCP error debugging.
  • debug sccp events —Displays SCCP events debugging.
  • debug sccp packets —Displays SCCP packets debugging.
  • debug sccp parser —Displays SCCP parser and builder debugging.

Use the following commands on the voice gateway to capture and view a log of STCAPP events:

  • debug voip application stcapp buffer-history —Enables event logging for STCAPP ports.
  • show stcapp buffer-history —Displays call flow and device events saved to the event log.

Use the following command on the voice gateway to filter output for debug commands based on the individual voice port:

  • debug condition voice-port port

Configuration Examples for Configuring FXS Ports for Basic Calls

This section contains the following examples:

Example: Cisco IOS Gateway SCCP Analog Ports Configuration

The following example shows a configuration for a Cisco VG224 Analog Phone Gateway in Cisco IOS Release 12.4(2)T:

Router# show running-config
 
Building configuration...
 
Current configuration : 3442 bytes
!
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource manager
!
ip subnet-zero
no ip dhcp use vrf connected
!
!
no ftp-server write-enable
!
stcapp ccm-group 1
stcapp
!
stcapp feature access-code
prefix *#
call forward all 2
call forward cancel 9
!
stcapp feature speed-dial
prefix ##
redial 9
voicemail 8
speed dial from 3 to 7
!
!
template address
!
voice-card 0
!
!
!
!
!
interface FastEthernet0/0
ip address 10.4.138.5 255.255.0.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip default-gateway 10.4.0.1
ip classless
ip route 223.255.254.0 255.255.255.0 10.4.0.1
!
ip http server
!
!
!
control-plane
!
!
voice-port 2/0
caller-id enable
!
voice-port 2/1
caller-id enable
!
voice-port 2/2
caller-id enable
!
voice-port 2/3
caller-id enable
!
voice-port 2/4
!
.
.
.
!
voice-port 2/23
!
!
!
sccp local FastEthernet0/0
sccp ccm 10.4.131.200 identifier 7815
sccp ccm 10.4.138.77 identifier 7825
sccp
!
sccp ccm group 1
associate ccm 7815 priority 1
associate ccm 7825 priority 2
!
!
dial-peer voice 500 pots
service stcapp
port 2/0
!
dial-peer voice 501 pots
service stcapp
port 2/1
!
dial-peer voice 502 pots
service stcapp
port 2/2
!
dial-peer voice 503 pots
service stcapp
port 2/3
!
dial-peer voice 504 pots
service stcapp
port 2/4
!
.
.
.
!
dial-peer voice 523 pots
service stcapp
port 2/23
!
!
line con 0
exec-timeout 0 0
transport preferred all
transport output all
line aux 0
transport preferred all
transport output all
line vty 0 4
login
transport preferred all
transport input all
transport output all
!
end

Example: PLAR with DTMF Out-Pulse Digits

The following example shows PLAR with DTMF Out-Pulse Digits configured on a Cisco VG224 voice gateway:

Router# show running-config
 
Building configuration...
 
Current configuration : 3442 bytes
.
.
.
!
stcapp ccm-group 1
stcapp
!
stcapp feature access-code
prefix *#
call forward all 2
call forward cancel 9
!
stcapp feature speed-dial
digit 2
voicemail 55
speed dial from 11 to 17
!
!
template address
!
voice-card 0
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 10.4.138.5 255.255.0.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip default-gateway 10.4.0.1
ip classless
ip route 223.255.254.0 255.255.255.0 10.4.0.1
!
ip http server
!
!
!
control-plane
!
!
voice-port 2/0
caller-id enable
!
voice-port 2/1
caller-id enable
!
voice-port 2/2
caller-id enable
!
voice-port 2/3
caller-id enable
!
voice-port 2/4
!
.
.
.
!
voice-port 2/23
!
!
!
sccp local FastEthernet0/0
sccp ccm 10.4.131.200 identifier 7815
sccp ccm 10.4.138.77 identifier 7825
sccp ccm 172.16.18.4 identifier 37454
sccp
!
sccp ccm group 1
associate ccm 7815 priority 1
associate ccm 7825 priority 2
associate ccm 37454 priority 3
registration timeout 3
keepalive retries 1
keepalive timeout 3
switchback method graceful
!
sccp plar
voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200
voiceport 2/1 dial 3264 digit 678,,,9*0,,#123 interval 100
voiceport 2/3 dial 3478 digit 34567 wait-connect 500
!
!
dial-peer voice 500 pots
service stcapp
port 2/0
!
dial-peer voice 501 pots
service stcapp
port 2/1
!
dial-peer voice 502 pots
service stcapp
port 2/2
!
dial-peer voice 503 pots
service stcapp
port 2/3
!
dial-peer voice 504 pots
service stcapp
port 2/4
!
.
.
.
!
dial-peer voice 523 pots
service stcapp
port 2/23
!
!
line con 0
exec-timeout 0 0
transport preferred all
transport output all
line aux 0
transport preferred all
transport output all
line vty 0 4
login
transport preferred all
transport input all
transport output all
!
end

Additional References

The following sections provide references related to SCCP analog phone support for FXS ports on the Cisco voice gateway.

Related Documents

Related Topic
Document Title

Cisco Unified Communications Manager

Cisco Unified Communications Manager documentation

Cisco Unified Communications Manager Express

Cisco Unified Communications Manager Express documentation

Cisco IOS debugging

Cisco IOS Debug Command Reference

Cisco IOS voice commands

Cisco IOS Voice Command Reference

Cisco IOS voice configuration

Cisco IOS Voice Configuration Library

Cisco voice gateway

Conferencing and transcoding resources

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Feature Information for Configuring FXS Ports for Basic Calls

Table 1 lists the features in this module and provides links to specific configuration information. Only features that were introduced or modified in Cisco IOS Release 12.4(6)XE or a later release appear in the table.

For information on a feature in this technology that is not documented here, see the “Supplementary Services Features Roadmap” section.

Not all commands may be available in your Cisco IOS software release. For release information about a specific command, see the command reference documentation.

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

note.gif

Noteblank.gif Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS software release train also support that feature.

 

Table 1 Feature Information for Configuring FXS Ports for Basic Calls

Feature Name
Releases
Feature Information

Hookflash Duration

12.4(6)XE
12.4(11)T

Enables modification of the upper limit of the hookflash duration range for an analog FXS port.

The following sections provide information about this feature:

No new commands were introduced by this feature.

SCCP Dial Tone Generation After Remote On-Hook

12.4(6)XE
12.4(11)T

Enables the SCCP gateway to generate dial tone to the remaining party in basic call mode once the remote party disconnects.

The following sections provide information about this feature:

The following commands were introduced or modified by this feature: tone dialtone remote-onhook

SCCP Gateway Ground Start FXS Ports

Supports ground-start FXS ports for disconnect supervision and Cisco Communications Manager registration of PBX and key system connections.

The following sections provide information about this feature:

No new commands were introduced by this feature.

SCCP PLAR with DTMF Out-Pulse Digits for FXS Analog Phones

12.4(6)T

Adds private line automatic ring-down (PLAR) support for SCCP analog ports on a Cisco VG224 Analog phone gateway.

The following sections provide information about this feature:

The following commands were introduced or modified by this feature: sccp plar, voiceport.

12.4(9)

Adds supports for FXS ports on Cisco ISRs.

Supervisory Disconnect

12.4(6)T

Provides a disconnect indication to the remote party in a two-party call after one side disconnects and enables external applications connected to the Cisco voice gateway to promptly clear a call after receiving the disconnect indication.

The following sections provide information about this feature:

The following commands were introduced or modified by this feature: supervisory disconnect, timeouts power-denial.

12.4(9)T

Adds supports for FXS ports on Cisco ISRs.