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Table Of Contents
FXO Answer and Disconnect Supervision
Supported Standards, MIBs, and RFCs
Configuring Voice Ports to Detect Supervisory Disconnect Tones
Configuring a Voice Class for FXO Supervisory Disconnect Tone
Assigning an FXO Supervisory Disconnect Tone Voice Class to an FXO Voice Port
Configuring an FXO Voice Port to Disconnect with any Detected Tone
Verifying Configuration of Disconnect Tone Detection Parameters
supervisory disconnect anytone
supervisory disconnect dualtone voice-class
FXO Answer and Disconnect Supervision
Document Update Alert
This document was originally produced for Cisco IOS Release 12.1(3)T and released as the FXO Supervisory Disconnect Tone feature. This feature has been updated in subsequent releases, and more recent documentation is available.
If you are using Cisco IOS Release 12.2(2)T or higher, refer to the following sections in the Voice Port Configuration Guide of the Cisco IOS Voice Configuration Library, Release 12.3:
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Configuring Disconnect Supervision
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If you are using Cisco IOS Release 12.1(3)T or higher, refer to the following sections in the Configuring Voice Ports chapter of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2:
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This feature module describes the FXO Supervisory Disconnect Tone feature for analog FXO ports with loop start signaling, introduced in Cisco IOS Release 12.1(3)T on the Cisco 2600 and 3600 series routers and Cisco MC3810 series concentrators.
This document includes the following sections:
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Feature Overview
If the FXO Supervisory Disconnect Tone feature is configured and a detectable tone from the PSTN or PBX is detected by the digital signal processor (DSP), the analog FXO port goes on-hook. This feature prevents an analog FXO port from remaining in an off-hook state after an incoming call is ended.
You can configure a voice port to detect either of the following tone types:
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You can configure the FXO Supervisory Disconnect Tone feature to function in either of the following ways:
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As part of the tone detection process by the DSP, a DSP event is reported to the host software.
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Detection of any tone is effective only during call set-up (before a call is answered), and echo cancellation must be enabled to prevent disconnection due to detection of the router's own ringback tone.
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Benefits
The FXO Supervisory Disconnect Tone feature allows interoperability with PSTN and PBX systems whether or not they transmit supervisory tones.
Restrictions
The FXO Supervisory Disconnect Tone feature is applicable only to analog FXO ports with loop-start signaling. Cadence detection is not supported.
The Cisco MC3810 series concentrators must be equipped with high-performance compression modules (HCMs) to support tone detection. Standard voice compression modules (VCMs) do not support the FXO Supervisory Disconnect Tone feature.
Related Documents
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Supported Platforms
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Supported Standards, MIBs, and RFCs
None
Prerequisites
The FXO Supervisory Disconnect Tone feature described in this document requires Cisco IOS Release 12.1(3)T or later.
Configuration Tasks
You can configure a voice port to disconnect when it detects a specific tone or tone pattern from a PBX or PSTN, or you can configure it to disconnect if it detects any tone before a call is answered.
For supervisory disconnect based on detection of a specific tone from a PBX or PSTN, complete the following tasks:
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For supervisory disconnect based on detection of any tone from a PBX or PSTN before a call is answered, complete the following tasks:
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Configuring Voice Ports to Detect Supervisory Disconnect Tones
To enable detection of supervisory disconnect tones, first create a voice class that defines the tone detection parameters, and then apply the voice class to the applicable analog FXO voice ports.
Configuring a Voice Class for FXO Supervisory Disconnect Tone
To configure a voice port to detect incoming tones, you need to know the parameters of the tones expected from the PBX or PSTN. This procedure configures the voice port to go on hook when it detects the specified tones. The parameters of the tones need to be precisely specified to prevent unwanted disconnects due to detection of non-supervisory tones or noise.
A supervisory disconnect tone is normally a dual tone consisting of two frequencies; however, tones of only one frequency can also be detected.
Caution
To create a voice class that defines the specific tone or tones to be detected, complete the following steps beginning in global configuration mode:
Assigning an FXO Supervisory Disconnect Tone Voice Class to an FXO Voice Port
To assign an FXO supervisory disconnect tone voice class to an analog FXO voice port, complete the following steps beginning in global configuration mode:
Step 1
For Cisco 2600 and 3600 series analog voice ports:
Router(config)# voice-port slot/subunit/portFor Cisco MC3810 series analog voice ports:
Router(config)# voice-port slot/portIdentify the voice port you want to configure and enter voice-port configuration mode.
Step 2
Assign an FXO supervisory disconnect tone voice class to the voice port. The voice class is one that you created in the "Configuring a Voice Class for FXO Supervisory Disconnect Tone" section.
Specify mid-call for tone detection during the entire call.
Specify pre-connect for tone detection only during call set-up.
Step 3
Router(config-voiceport)# exitExit from voice-port configuration mode.
Configuring an FXO Voice Port to Disconnect with any Detected Tone
To configure an analog FXO voice port to go on-hook upon receipt of any tone received from a PBX or PSTN before the call is answered, complete the following steps beginning in global configuration mode:
Verifying Configuration of Disconnect Tone Detection Parameters
Use either or both of the following methods to verify that the tone detection parameters have been properly configured on a voice port:
Use the "show running-config" Command
Enter the show running-config command to review the tone detection parameters you have configured.
The following example shows part of the output from the show running-config command on a Cisco MC3810, in which voice class 70 defines the tone parameters:
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router# show running-configBuilding configuration......voice class dualtone 70freq-pair 1 350 440freq-pair 2 500 800...voice-port 1/6supervisory disconnect dualtone mid-call voice-class 70...Additional display lines are not shown.
Send a Supervisory Disconnect Tone from a PBX, the PSTN, or a SAGE Call Generator
Step 1
Step 2
Note
If you use a SAGE, the tone generated by the SAGE must match the parameters you have configured in the router for supervisory disconnect.Step 3
The following is a sample display from the show voice port summary command on a Cisco MC3810, in which analog FXO voice port 1/2 is on hook:
router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC====== == ========== ===== ==== ======== ======== ==1/2 -- fxo-ls up dorm idle on-hook y
Configuration Examples
Applying a Voice Class to a Voice Port
This example applies voice class 70 to FXO voice port 1/2 on a Cisco MC3810, and specifies tone detection for the entire duration of the call.
Router(config)# voice port 1/1Router(config-voiceport)# no echo-cancel enableRouter(config-voiceport)# connection plar 12Router(config-voiceport)# supervisory disconnect dualtone mid-call voice-class 70Router(config-voiceport)# exitThis example applies voice class 80 to FXO voice port 0/1/0 on a Cisco 3600 series router, and specifies tone detection for the call set-up time only.
Router(config)# voice port 0/1/0Router(config-voiceport)# no echo-cancel enableRouter(config-voiceport)# connection plar 12Router(config-voiceport)# supervisory disconnect dualtone pre-connect voice-class 80Router(config-voiceport)# exitConfiguring a Voice Port to Disconnect with Any Detected Tone
This example configures voice port 1/1 to go on-hook upon receipt of any tone from a PBX or PSTN if the caller goes on-hook before the call is answered:
Router(config)# voice port 1/1Router(config-voiceport)# echo-cancel enableRouter(config-voiceport)# connection plar 12Router(config-voiceport)# supervisory disconnect anytoneRouter(config-voiceport)# exitCommand Reference
This section documents new commands. All other commands used on these platforms are documented in the Cisco IOS Release 12.1 command reference publications.
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freq-pair
To specify the frequency components of a tone to be detected, use the freq-pair voice-class command. To cancel detection of a tone, use the no form of this command.
freq-pair tone-id frequency-1 frequency-2
no freq-pair tone-id
Syntax Description
Defaults
No tone is specified for detection.
Command Modes
Voice-class configuration.
Command History
12.1(3)T
This command was introduced on the Cisco 2600, 3600, and MC3810 series.
Usage Guidelines
To detect a tone with two frequency components (a dualtone), configure frequencies for frequency-1 and frequency-2.
To detect a tone with only one frequency component, configure a frequency for frequency-1 and enter 0 for frequency-2.
You can configure a router to detect up to 16 tones.
Examples
The following example configures tone number 1 (tone-id 1) with frequency components of 480 and 2400 Hz:
Router(config)# voice class dualtone 100Router(config-voice-class)# freq-pair 1 480 2400Router(config-voice-class)# exitThe following example configures tone number 1 (tone-id 1) with frequency components of 480 Hz and 2400 Hz, and tone number 2 (tone-id 2) with frequency components of 560 Hz and 880 Hz:
Router(config)# voice class dualtone 50Router(config-voice-class)# freq-pair 1 480 2400Router(config-voice-class)# freq-pair 2 560 880Router(config-voice-class)# exitRelated Commands
supervisory disconnect anytone
To configure an FXO voice port to go on-hook if the router detects any tone from a PBX or PSTN before the call is answered, use the supervisory disconnect anytone voice-port command. To restore the default, use the no form of this command.
supervisory disconnect anytone
no supervisory disconnect anytone
Syntax Description
This command has no arguments or keywords.
Defaults
The supervisory disconnect function is not enabled on voice ports.
Command Modes
Voice-port configuration.
Command History
12.1(3)T
This command was introduced on the Cisco 2600, 3600, and MC3810 series.
Usage Guidelines
The supervisory disconnect anytone voice-port command can be used to provide the disconnect function in cases where the PBX or PSTN does not provide a supervisory tone.
This function is enabled only during call setup (before the call is answered); examples of tones that trigger a disconnect include busy tone, fast busy tone, and dial tone.
You must enable echo cancellation; otherwise, the router's own ringback tone can trigger a disconnect.
This command replaces the no supervisory disconnect signal command.
If you enter the no supervisory disconnect signal command, the supervisory disconnect anytone feature will be enabled, and supervisory disconnect anytone will be displayed when show commands are entered.
Examples
The following example configures voice ports 1/4 and 1/5 to go on hook if any tone from the PBX or PSTN is detected before the call is answered:
Router(config)# voice-port 1/4Router(config-voice-class)# supervisory disconnect anytoneRouter(config-voice-class)# exitRouter(config)# voice-port 1/5Router(config-voice-class)# supervisory disconnect anytoneRouter(config-voice-class)# exitThe following example disables the disconnect function on voice port 1/5:
Router(config)# voice-port 1/5Router(config-voice-class)# no supervisory disconnect anytoneRouter(config-voice-class)# exitRelated CommandsRelated Commands
supervisory disconnect dualtone voice-class
To assign a previously-configured voice class for FXO supervisory disconnect tone to a voice port, use the supervisory disconnect dualtone voice-class voice port configuration command. To remove a voice class from a voice port, use the no form of this command.
supervisory disconnect dualtone {mid-call | pre-connect} voice-class tag
no supervisory disconnect dualtone voice-class tag
Syntax Description
Defaults
No voice class is assigned to a voice port.
Command Modes
Voice-port configuration.
Command History
Usage Guidelines
You can apply an FXO supervisory disconnect tone voice class to multiple voice ports.
You can assign only one FXO supervisory disconnect tone voice class to a voice port. If a second voice class is assigned to a voice port, the second voice class replaces the one previously assigned.
You cannot assign separate FXO supervisory disconnect tone commands directly to the voice port.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example assigns voice class 70 to FXO voice port 1/5 of a Cisco MC3810 series concentrator, and specifies tone detection during the entire call duration:
Router(config)# voice-port 1/5Router(config)# no echo-cancel enableRouter(config-voiceport)# supervisory disconnect dualtone mid-call voice-class 70Router(config-voiceport)# exitThe following example assigns voice class 80 to FXO voice port 0/1/1 of a Cisco 3600 series router, and specifies tone detection only during call set-up:
Router(config)# voice-port 0/1/1Router(config)# no echo-cancel enableRouter(config-voiceport)# supervisory disconnect dualtone pre-connect voice-class 80Router(config-voiceport)# exitRelated CommandsRelated Commands
voice class dualtone
To create a voice class for FXO supervisory disconnect tone detection parameters, use the voice class dualtone global configuration command. To delete the voice class, use the no form of this command.
voice class dualtone tag
no voice class dualtone tag
Syntax Description
Defaults
No voice class is configured for tone detection parameters.
Command Modes
Global configuration.
Command History
12.1(3)T
This command was introduced.
12.2(2)T
This command was replaced by the
voice-class dualtone-detect-params and the
voice-class custom-cptone commands.
Usage Guidelines
Use this command first to create the voice class. Then use the supervisory disconnect dualtone voice-class command to assign the voice class to a voice port.
A voice class can define any number of tones to be detected. You need to define a matching tone for each supervisory disconnect tone expected from a PBX or from the PSTN.
Examples
The following example configures voice class dualtone 70, which defines one tone with two frequency components, and does not configure a cadence list:
Router(config)# voice class dualtone 70Router(config-voice-class)# freq-pair 1 350 440Router(config-voice-class)# freq-max-deviation 10Router(config-voice-class)# freq-max-power 6Router(config-voice-class)# freq-min-power 25Router(config-voice-class)# freq-power-twist 15Router(config-voice-class)# freq-max-delay 16Router(config-voice-class)# exitThe following example configures voice class dualtone 100, which defines one tone with two frequency components, and configures a cadence list:
Router(config)# voice class dualtone 70Router(config-voice-class)# freq-pair 1 350 440Router(config-voice-class)# freq-pair 2 480 850Router(config-voice-class)# freq-max-deviation 10Router(config-voice-class)# freq-max-power 6Router(config-voice-class)# freq-min-power 25Router(config-voice-class)# freq-power-twist 15Router(config-voice-class)# freq-max-delay 16Router(config-voice-class)# exitThe following example configures voice class dualtone 90, which defines three tones, each with two frequency components, and configures two cadence lists:
Router(config)# voice class dualtone 90Router(config-voice-class)# freq-pair 1 350 440Router(config-voice-class)# freq-pair 2 480 850Router(config-voice-class)# freq-pair 3 1000 1250Router(config-voice-class)# freq-max-deviation 10Router(config-voice-class)# freq-max-power 6Router(config-voice-class)# freq-min-power 25Router(config-voice-class)# freq-power-twist 15Router(config-voice-class)# freq-max-delay 16Router(config-voice-class)# exitRelated CommandsRelated Commands
Assigns a previously-configured voice class for FXO supervisory disconnect tone to a voice port.
Glossary
ABCD signaling—Four-bit telephony line signaling coding in which each letter of "ABCD" represents one of the four bits. This is often associated with CAS or Robbed-Bit signaling on a T1 or E1 telephony trunk.
AIS—Alarm Indication Signal.
AVBO—Advanced Voice Busy Out.
Cisco trunk (private line) call—A Cisco trunk (private line) call is established by the forced connection of a dynamic switched call. A Cisco trunk call is established during configuration of the trunk and stays up for the duration of the configuration. Optionally, it provides a pass-through connection path to pass signaling information between the two telephony interfaces at either end of the connection.
CLI—Command line interface.
codec—coder-decoder. An integrated circuit device that typically uses pulse code modulation to transform analog signals into a digital bit stream, and digital signals back into analog signals. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/decompress speech or audio signals.
DLCI—Data-link connection identifier.
Dial peer—An addressable call endpoint that contains configuration information including voice protocol, a CODEC type, and a telephone number associated with the call endpoint. There are five kinds of dial peers: POTS, VoIP, VoFR, VoATM, and VoHDLC.
DSP—Digital Signaling Processor.
DTMF—Dual tone multi frequency. Uses two simultaneous voice-band tones for dial such as touch tone.
DTMF relay—Enables the generation of FRF.11 Annex A frames for a VoFR dial peer. The DSP generates Annex A frames instead of passing a DTMF tone through the network as a voice sample.
Dynamic switched call—A telephone call dynamically established across a packet data network based on a dialed telephone number. In the case of VoFR, a Cisco proprietary session protocol similar to Q.931 is used to achieve call switching and negotiation between calling endpoints. The proprietary session protocol runs over FRF.11-compliant subchannels.
E&M—Stands for 2-wire or 4-wire interfaces with separate signaling paths (from "Ear and Mouth," also "recEive and transMit"). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. The Cisco analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.
E1—European equivalent of T1. 32-64kbps channels include 1-channel for framing and 1-channel for D-channel information. The clock rate is 2.048 Mhz.
FRF—Frame Relay Forum. An association of corporate members consisting of vendors, carriers, users, and consultants committed to implementing Frame Relay in accordance with national and international standards. See http://www.frforum.com.
FRF.11—Frame Relay Forum implementation agreement for Voice over Frame Relay (v1.0 May 1997). This specification defines multiplexed data, voice, fax, DTMF digit-relay, and CAS/Robbed-bit signaling frame formats, but does not include call setup, routing, or administration facilities. See http://www.frforum.com.
FRF.11 Annex C—See FRF.12.
FRF11-trunk—A point-to-point permanent voice connection (private line) conforming to the FRF.11 specification.
FRF.12—The FRF.12 Implementation Agreement (also known as FRF.11 Annex C) was developed to allow long data frames to be fragmented into small pieces and interleaved with real-time frames. In this way, real-time voice and non real-time data frames can be carried together on low speed links without causing excessive delay to the real-time traffic.
FXO—Foreign Exchange Office. An FXO interface connects to the Public Switched Telephone Network (PSTN) central office and is the interface offered on a standard telephone. The Cisco FXO interface is an RJ-11 connector that allows an analog connection to be directed to the PSTN central office or to a station interface on a PBX.
FXS—Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. The Cisco FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.
ICPIF—Calculated Planning Impairment Factor.
LVBO—Local Voice Busy Out.
MEL CAS—Mercury Exchange Limited (MEL) Channel Associated Signaling (CAS). A voice signaling protocol used primarily in the United Kingdom.
OOS—Out of Service state of the call or trunk.
PBX—Private Branch Exchange. A privately owned central switching office.
Permanent calls—Permanent calls are private line calls used for fixed point-to-point calls, connections between PBXs (E&M to E&M), or for remote telephone extensions (FXO to FXS).
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
POTS dial peer—Dial peer connected by a traditional telephony network. POTS peers point to a particular voice port on a voice network device.
PSTN—Public Switched Telephone Network. PSTN refers to the local telephone company.
RTR—Response Time Reporter.
Switched calls—Switched calls are normal telephone calls when a user picks up a phone, hears a dial tone, and enters the destination phone number to reach the other phone. Switched calls can also be private line auto-ringdown (PLAR) calls, or tie-line calls for fixed point-to-point connections.
T1—Digital WAN carrier facility. T1 transmits DS-1-formatted data at 1.544 Mbps through the telephone-switching network by using AMI or B8ZS coding.
Tandem switching—The dynamic switching of voice calls between VoFR, VoATM, or VoHDLC PVCs and subchannels; also called tandeming. Tandem switching is often encountered in multi-hop VoFR call connection paths.
Trunk—Service that allows quasi-transparent connections between two PBXs, a PBX and a local extension, or some other combination of telephony interfaces with signaling passed transparently through the packet data network.
VoFR—Voice over Frame Relay.
VoFR dial peer—Dial peer connected by a Frame Relay network. VoFR peers point to specific VoFR devices.
Voice over Frame Relay—Voice over Frame Relay enables a router to carry voice traffic, for example, telephone calls and faxes, over a Frame Relay network. When sending voice traffic over Frame Relay, the voice traffic is segmented and encapsulated for transit across the Frame Relay network by using FRF.12 encapsulation.
Voice over IP—Voice over IP enables a router to carry voice traffic, for example, telephone calls and faxes, over an IP network. In Voice over IP, the DSP segments the voice signal into frames, which are then coupled in groups of two and stored in voice packets that are transported by using IP in compliance with ITU-T specification H.323.
VoIP—Voice over IP through Ethernet.
