Table of Contents

Benefits
Restrictions
Related Documents

Supported Platforms
Prerequisites
Configuration Tasks

Configuring Voice Ports

Configuring FXO or FXS Voice Ports
Fine-Tuning FXO and FXS Voice Ports
Configuring E&M Voice Ports
Fine-Tuning E&M Voice Ports
Activating the Voice Port

Configuring ATM Interfaces to Support Voice Traffic
Preparing to Configure Voice Dial Peers
Creating a Peer Configuration Table
Configuring Dial Peers

Configuring POTS Dial Peers
Configuring Voice over ATM Dial Peers

Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router

Verifying the Voice Connection
Troubleshooting Tips

Voice over ATM Configuration Example
Command Reference
dial-peer hunt
dial-peer terminator
dial-peer voice
encapsulation
forward-digits
huntstop
session target
signal-type
vbr-rt
voice hunt user-busy
Debug Commands
debug ccswvoice voatm-debug
debug ccswvoice voatm-session
Glossary

Voice over ATM on
Cisco 3600 Series Routers

Voice over ATM on Cisco 3600 series routers extends support for Voice over ATM, previously available only on the Cisco MC3810 to the Cisco 3600 series routers. Voice over ATM enables a Cisco 3600 series router to carry voice traffic (for example, telephone calls and faxes) over an ATM network.

Benefits

Voice over ATM enables a Cisco 3600 to carry voice traffic, (for example, telephone calls and faxes) over an ATM network by using ATM encapsulation AAL5.

Restrictions

• Voice over ATM on the Cisco 3600 series requires that you install one of the following modules:

  • Multiport T1/E1 ATM network module with IMA

  The Multiport T1/E1 ATM network module with IMA supports up to 8 T1/E1 lines.
  

  For more information, see the Cisco IOS Release 12(05)T online document Configuring Multiport T1/E1 ATM Network Modules with Inverse Multiplexing over ATM on Cisco 2600 and 3600 Series Routers.
  

  • OC3 ATM Network Module

  The OC3 ATM Network Module supports one OC3 line.
  

  For more information about the Digital T1 packet voice trunk network modules, see the Cisco IOS Release 12.0(3)T online document ATM OC-3 Network Module for the Cisco 3600 Series Routers.
  

• Voice over ATM on the Cisco 3600 series supports ATM encapsulation AAL5 only. AAL2 is not supported.

• Voice over ATM Switched Virtual Circuits (SVCs) are not supported in this release.

Related Documents

• For information about voice port changes in Cisco IOS Release 12.1(2)T, see the following Cisco IOS 12.1(2)T online documents:

  • Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators

  • Voice Port Testing Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators

  • Voice Busyout Enhancements on the Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators

• For information about software configuration requirements for the Digital T1 Packet Voice trunk network modules on the Cisco 2600 and Cisco 3600, see the Cisco IOS 12.1(2)T online document Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers.

For more information about voice technologies, see the Cisco IOS Multiservice Applications Configuration Guide, and the Cisco IOS Multiservice Applications Command Reference for the Cisco IOS Release 12.1.

Supported Platforms

• Cisco 3620

• Cisco 3640

• Cisco 3660

Note Voice over ATM on the Cisco 3600 series requires special voice and network modules to be installed. For more information, see the "Restrictions" section.

Prerequisites

Before you can configure your router to use Voice over ATM, you must first:

• Install the required network modules into the Cisco 3600 series router. For more information, see the "Restrictions" section.

The Multiport T1/E1 ATM network module with IMA supports up to 8 T1/E1 lines. The OC3 ATM Network Module supports one OC3 line.

• Establish a working ATM network. For more information about configuring ATM, see the Cisco IOS 12.1 Wide Area Networking Configuration Guide.

• Complete your company's dial plan.

• Establish a working telephony network based on your company's dial plan.

  • Integrate your dial plan and telephony network into your existing ATM network topology. Make routing and dialing transparent to the user---for example, avoid secondary dial tones from secondary switches where possible.

  • Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX interfaces.

After you analyze your dial plan and decide how to integrate it into your existing ATM network, you are ready to configure your network devices to support Voice over ATM.

Configuration Tasks

This section describes the following tasks required to configure Voice over ATM on the Cisco 3600 series routers:

• Configuring Voice Ports

• Configuring ATM Interfaces to Support Voice Traffic

• Preparing to Configure Voice Dial Peers

• Configuring Dial Peers

• Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router

Configuring Voice Ports

This section describes how to configure voice ports to support Voice over ATM. For more information about voice-port enhancements in Cisco IOS Release 12.1(2)T, see the online document Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators.

This section is divided into the following procedures:

• Configuring FXO or FXS Voice Ports

• Fine-Tuning FXO and FXS Voice Ports

• Configuring E&M Voice Ports

• Fine-Tuning E&M Voice Ports

• Activating the Voice Port

Voice ports provide support for three basic voice signaling formats:

• FXO---The Foreign Exchange Office interface. This interface allows a connection to be directed to the PSTN's central office. The FXO interface also allows a connection to be directed to a standard PBX interface if the local telecommunications authority permits. This interface is of value for off-premise extension applications.

• FXS---The Foreign Exchange Station interface. This interface allows connection for basic telephone equipment and keysets, and supplies ring, voltage, and dial tone.

• E&M---The "Ear and Mouth" (or "RecEive and TransMit") interface allows connection for PBX trunk lines (tie lines). E&M is a signaling technique for two-wire and four-wire telephone and trunk interfaces.

In general, voice-port commands define the characteristics associated with a particular voice-port signaling type. Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.

The Cisco 3600 Series routers provide analog and digital voice ports. The type of signaling associated with these analog voice ports depend on the interface module installed into the device. The Cisco 3600 series router supports either a two-port or four-port voice network module (VNM); VNMs can hold either two or four voice interface cards (VICs).

Each VIC is specific to a particular signaling type; therefore, VICs determine the type of signaling for the voice ports on that particular VNM. This means that even though VNMs can hold multiple VICs, each VIC on a VNM must conform to the same signaling type. For more information about the physical characteristics of VNMs and VICs or how to install them, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.

Configuring FXO or FXS Voice Ports

Under most circumstances the default voice-port values are adequate for both FXO and FXS voice ports. To configure FXO and FXS voice ports, use the following commands beginning in global configuration mode:

	Command	Purpose
Step 1	For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group	Identify the voice port you want to configure and enter voice-port configuration mode.
Step 2	Router(config-voiceport)# dial-type {pulse | dtmf}	(For FXO ports only) Select the appropriate dial type for out-dialing. The default is dtfm.
Step 3	Router(config-voiceport)# signal {loop-start | ground-start}	Configure the signaling type for analog FXO and FXS voice ports. The default is loop-start.
Step 4	Router(config-voiceport)# compand-type {u-law | a-law}	Configure the companding standard used to convert between analog and digital signals in PCM systems. This command applies to digital voice-ports only.
Step 5	Router(config-voiceport)# cptone locale	Configure the appropriate call progress tone for the local region. The default for this command is us.
Step 6	Router(config-voiceport)# ring number number	(For FXO ports only) Configure the number of rings detected before a connection is closed on the FXO port.
Step 7	Router(config-voiceport)# ring frequency number	(For FXS ports only) Specify the local ring frequency for the FXS voice port.
Step 8	Router(config-voiceport)# music-threshold number	(Optional) Specify the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30.
Step 9	Router(config-voiceport)# no battery reversal	(Optional) Disable battery polarity reversal on an FXO or FXS port. This command is enabled by default. For more information, see the Cisco IOS Release 12.1(2)T document Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators.
Step 10	Router(config-voiceport)# no disconnect-ack	(Optional) Configure an FXS voice port to not return an acknowledgment upon receipt of a disconnect signal. This command is enabled by default.
Step 11	Router(config-voiceport)# connection {plar |  tie-line | plar-opx} string	(Optional) Configure the voice-port connection mode type and the destination telephone number. The plar value is used for Private Line Auto Ringdown (PLAR) connections. The tie-line value on the Cisco MC3810 is used for a tie-line connection to a PBX. The plar-opx value is used for PLAR Off-Premises eXtension, to allow the local voice port to provide a local response before the remote voice port receives an answer. Note If you will be configuring Cisco-trunk permanent connections, do not configure this command now. See the "Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router" section.
Step 12	Router(config-voiceport)# description string	(Optional) Enter a string description for the voice port. The string describes the voice port in displays. You can use the description command to note the voice port's location or use.
Step 13	Router(config-voiceport)# voice vad-time milliseconds	(Optional) Change the minimum silence detection time for voice activity detection (VAD). The range is 250 to 65536. The default is 250.
Step 14	Router(config-voiceport)# comfort-noise	(Optional) Specify that background noise will be generated if you have VAD activated.

Verifying Tips

You can check the validity of your voice-port configuration by performing the following tasks:

If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:

Fine-Tuning FXO and FXS Voice Ports

Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain and output attenuation for FXO or FXS voice ports. Collectively, these commands are referred to as voice-port tuning commands.

Note In most cases, the default values for voice-port tuning commands will be sufficient.

To fine-tune FXO or FXS voice ports, perform the following steps beginning in global configuration mode:

	Command	Purpose
Step 1	For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group	Identify the voice port you want to configure and enter voice-port configuration mode.
Step 2	Router(config-voiceport)# input gain value	Specify (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14.
Step 3	Router(config-voiceport)# output attenuation value	Specify (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14.
Step 4	Router(config-voiceport)# echo-cancel enable	Enable echo-cancellation of voice that is sent out the interface and received back on the same interface.
Step 5	Router(config-voiceport)# echo-cancel coverage value	Adjust the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32.
Step 6	Router(config-voiceport)# timeouts initial seconds	Configure the initial timeout value. The initial timeout value specifies the number of seconds the system waits for the caller to input the first digit of the dialed digits. The default is 10 seconds.
Step 7	Router(config-voiceport)# timeouts interdigit seconds	Configure the interdigit timeout value. The timeouts interdigit value specifies the number of seconds the system waits (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The default is 10 seconds.
Step 8	Router(config-voiceport)# timeouts ringing {seconds | infinity}	Configure the timeout value for ringing. The range is 5 to 60000. The default is 180.
Step 9	Router(config-voiceport)# timeouts wait-release {seconds | infinity}	Configure the delay timeout before the system starts the process for releasing voice ports. The range is 3 to 3600. The default is 30.
Step 10	Router(config-voiceport)# timing digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50 to 100 milliseconds. The default is 100.
Step 11	Router(config-voiceport)# timing guardout milliseconds	(FXO ports only) Specify the guard-out duration of the FXO voice port. The range is 300 to 3000. The default is 2000.
Step 12	Router(config-voiceport)# timing percentbreak percent	(FXO ports only) Specify the percentage of the break period for dialing pulses for a voice port. Valid entries are numbers 20 to 80. The default is 50.
Step 13	Router(config-voiceport)# timing inter-digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF inter-digit signal duration. The range of the DTMF inter-digit signal duration is from 50 to 500 milliseconds. The default is 100.
Step 14	Router(config-voiceport)# timing pulse-digit milliseconds	If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from10 to 20 milliseconds. The default is 20.
Step 15	Router(config-voiceport)# timing pulse-inter-digit milliseconds	If the voice-port dial type is pulse, configure the pulse inter-digit signal duration. The range of the pulse inter-digit signal duration is from 100 to 1000 milliseconds. The default is 500.
Step 16	Router(config-voiceport)# impedance {600r | 600c | 900r | 900c}	(For FXO ports only) Configure the impedance. The default is 600 ohms real.
Step 17	Router(config-voiceport)# ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]}	(For FXS ports only) Specify the ring cadence for the FXS voice port.

Note After you change any voice-port command, cycle the port by using the shutdown and no shutdown commands.

Configuring E&M Voice Ports

Unlike FXO and FXS voice ports, the default E&M voice-port parameters most likely will not be sufficient to enable voice data transmission over your network.

Note E&M voice-port values must match those of the PBX to which it is connected. Refer to the documentation that came with your specific PBX for the appropriate E&M voice-port configuration command values.

To configure E&M voice ports, use the following commands beginning in global configuration mode:

	Command	Purpose
Step 1	For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group	Identify the voice port you want to configure and enter voice-port configuration mode.
Step 2	Router(config-voiceport)# dial-type {dtmf | pulse}	Select the appropriate dial type for out-dialing.
Step 3	Router(config-voiceport)# operation {2-wire | 4-wire}	Select the appropriate cabling scheme for this voice port.
Step 4	Router(config-voiceport)# type {1 | 2 | 3 | 5}	Select the appropriate E&M interface type. Type 1 indicates the following lead configuration: E---output, relay to ground M---input, referenced to ground Type 2 indicates the following lead configuration: E---output, relay to SG M---input, referenced to ground SB---feed for M, connected to -48V SG---return for E, galvanically isolated from ground Type 3 indicates the following lead configuration: E---output, relay to ground M---input, referenced to ground SB---connected to -48V SG---connected to ground Type 5 indicates the following lead configuration: E---output, relay to ground M---input, referenced to -48V.
Step 5	Router(config-voiceport)# signal {wink-start | immediate | delay-dial}	Configure the signaling type for E&M voice ports. The default is wink-start.
Step 6	Router(config-voiceport)# impedance {600c | 600r | 900c | complex1 | complex2}	Specify a terminating impedance. This value must match the specifications from the telephony system to which this voice port is connected.
Step 7	Router(config-voiceport)# connection {plar |  tie-line | plar-opx} string	(Optional) Configure the voice-port connection mode type and the destination telephone number. The plar value is used for Private Line Auto Ringdown (PLAR) connections. The tie-line value is used for a tie-line connection to a PBX. The plar-opx value is used for PLAR Off-Premises eXtension, to allow the local voice port to provide a local response before the remote voice port receives an answer. Note If you will be configuring Cisco-trunk permanent connections, do not configure this command now. See the "Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router" section.
Step 8	Router(config-voiceport)# compand-type {u-law | a-law}	Configure the companding standard used to convert between analog and digital signals in PCM systems. This command applies to digital voice-posts only.
Step 9	Router(config-voiceport)# cptone locale	Configure the appropriate call progress tone for the local region. The default for this command is us.
Step 10	Router(config-voiceport)# music-threshold number	(Optional) Specify the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30.
Step 11	Router(config-voiceport)# no auto-cut-through	(Optional) Disables call completion when a PBX does not provide an M-lead response. This command is enabled by default.
Step 12	Router(config-voiceport)# impedance {600c | 600r | 900c | complex1 | complex2}	Specify a terminating impedance. This value must match the specifications from the telephony system to which this voice port is connected.

Verifying Tips

You can check the validity of your voice-port configuration by performing the following tasks:

If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:

Fine-Tuning E&M Voice Ports

Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain and output attenuation for E&M voice ports. Collectively, these commands are referred to as voice-port tuning commands.

Note In most cases, the default values for voice-port tuning commands will be sufficient.

To fine-tune E&M voice ports, use the following commands beginning in global configuration mode:

	Command	Purpose
Step 1	For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group	Identify the voice port you want to configure and enter voice-port configuration mode.
Step 2	Router(config-voiceport)# input gain value	Specify (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14.
Step 3	Router(config-voiceport)# output attenuation value	Specify (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14.
Step 4	Router(config-voiceport)# echo-cancel enable	Enable echo-cancellation of voice that is sent out the interface and received back on the same interface.
Step 5	Router(config-voiceport)# echo-cancel coverage value	Adjust the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32.
Step 6	Router(config-voiceport)# non-linear	Enable non-linear processing, which shuts off any signal if no near-end speech is detected. (Non-linear processing is used with echo-cancellation.)
Step 7	Router(config-voiceport)# timeouts initial seconds	Configure the initial timeout value. The initial timeout value specifies the number of seconds the system waits for the caller to input the first digit of the dialed digits. The default is 10 seconds.
Step 8	Router(config-voiceport)# timeouts interdigit seconds	Configure the interdigit timeout value. The timeouts interdigit value specifies the number of seconds the system waits (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The default is 10 seconds.
Step 9	Router(config-voiceport)# timeouts wait-release {value | infinity}	Configure the timeout value for releasing voice ports. This command limits the duration that a voice port stays in the call failure state while the Cisco MC3810 sends a busy tone, reorder tone or out-of-service tone to the port.
Step 10	Router(config-voiceport)# timing digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50 to 100 milliseconds. The default is 100.
Step 11	Router(config-voiceport)# timing inter-digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF inter-digit signal duration. The range of the DTMF inter-digit signal duration is from 50 to 500 milliseconds. The default is 100.
Step 12	Router(config-voiceport)# timing percentbreak percent	Specify the percentage of the break period for dialing pulses for a voice port. Valid entries are numbers 20 to 80. The default is 50.
Step 13	Router(config-voiceport)# timing pulse-digit milliseconds	If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from10 to 20 milliseconds. The default is 20.
Step 14	Router(config-voiceport)# timing pulse-inter-digit milliseconds	If the voice-port dial type is pulse, configure the pulse inter-digit signal duration. The range of the pulse inter-digit signal duration is from 100 to 1000 milliseconds. The default is 500.
Step 15	Router(config-voiceport)# timing wink-duration milliseconds	Configure the timing wink-duration value. This value sets the wink signal duration for a wink-start signal. This value applies only if the signal command is set to "wink-start." The range is from 100 to 400 milliseconds and the default is 200.
Step 16	Router(config-voiceport)# timing wink-wait milliseconds	Configure the timing wink-wait value. This value sets the wink wait duration for a wink-start signal. This value applies only if the signal command is set to "wink-start." The range is from 100 to 5000 milliseconds and the default is 200.
Step 17	Router(config-voiceport)# timing clear-wait milliseconds	Configure the timing clear-wait value. This value sets the amount of time between the inactive seizure signal and the call being cleared. The range is from 100 to 2000 milliseconds and the default is 400.
Step 18	Router(config-voiceport)# timing delay-duration milliseconds	Configure the timing delay-duration value. This value sets the delay signal duration for delay dial signaling. This value applies only if the signal command is set to "delay-dial." The range is from 100 to 5000 milliseconds and the default is 140.
Step 19	Router(config-voiceport)# timing delay-start milliseconds	Configure the timing delay-start value. This value sets the delay interval between the generation of the delay-start signal from incoming detection seizure. This value applies only if the signal command is set to "delay-dial." The range is from 100 to 290 milliseconds and the default is 150.
Step 20	Router(config-voiceport)# timing percentbreak percent	Configure the timing percent-break value. This value sets the percentage of the break period for a dialing pulse. The default is 50 percent.

Note After you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands.

Activating the Voice Port

After you have configured the voice port, you need to activate the voice port to bring it online. In fact it is a good idea to cycle the port---meaning to shut the port down and then bring it online again.

To activate a voice port, use the following command in voice-port configuration mode:

	Command	Purpose
Step 1	Router(config-voiceport)# no shutdown	Activate the voice port.

To cycle a voice port, use the following commands in voice-port configuration mode:

	Command	Purpose
Step 1	Router(config-voiceport)# shutdown	Deactivate the voice port.
Step 2	Router(config)# voice-port slot/subunit/port	Enter voice-port configuration mode.
Step 3	Router(config-voiceport)# no shutdown	Activate the voice port.

Note If you are not going to use a voice port, shut it down.

Configuring ATM Interfaces to Support Voice Traffic

This section describes the preliminary ATM configuration tasks necessary to support Voice over ATM.

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Note The Voice over ATM configuration must be performed on the routers on both sides of the voice connection.

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To configure the Cisco 3600 series to support Voice over ATM, use the following commands beginning in global configuration mode:

	Command	Purpose
Step 1	Router(config)# interface ATMslot/number	Enter ATM interface configuration mode.
Step 2	Router(config-if)# pvc [name] vpi/vci	Create an ATM PVC for voice traffic and enter virtual circuit configuration mode.
Step 3	Router(config-if-atm-pvc)# encapsulation aal5mux voice	Set the encapsulation of the PVC to support voice traffic. Note To configure a PVC to support data traffic, use aal5snap encapsulation.
Step 4	Router(config-if-atm-pvc)# vbr-rt peak-rate average-rate [burst]	Configure the peak rate, average rate, and the burst cell size to perform traffic shaping between voice and data PVCs. By using the vbr-rt command, you can configure the variable bit-rate for real-time networks, such as for voice networks. Traffic shaping is necessary, so that the carrier does not discard the incoming calls from the router. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will be carrying bursty traffic. The peak, average, and burst values are needed so the PVC can effectively handle the bandwidth for the number of voice calls. To calculate the minimum peak, average, and burst values for the number of voice calls, use the following calculations: Peak value: (2 x the maximum number of calls) x 16 kb The peak value equals the Peak Information Rate (PIR). Average value: (1 x the maximum number of calls) x 16  kb The average value equals the Average Information Rate (AIR). This correlates to the carrier's sustained cell rate. Burst value: (4 x the maximum number of calls) The burst value is the burst size in cells. Note When you configure data PVCs that will be traffic shaped with voice PVCs, use the aal5snap encapsulation and calculate the overhead as 1.13 times the voice rate.
Step 5	Router(config-if-atm-pvc)# exit	Exit ATM virtual circuit configuration mode. The only commands in ATM virtual circuit configuration mode used for ATM voice PVCs are encapsulation aal5mux voice, vbr-rt, and ilmi.
Step 6	Router(config-if)# pvc [name] vpi/vci	Create an ATM PVC for data traffic and enter virtual circuit configuration mode.
Step 7	Router(config-if-atm-pvc)# encapsulation aal5snap	Set the encapsulation of the PVC to support ATM data traffic. In ATM PVC configuration mode, configure either the ubr, ubr+, or the vbr-nrt traffic shaping commands for the data PVC as appropriate.
Step 8	Router(config-if-atm-pvc)# exit	Exit ATM virtual circuit configuration mode. Repeat Steps 6 and 7 for each data PVC configured.
Step 9	Router(config-if)# exit	Exit interface configuration mode.
Step 10	Router(config)# exit	Exit configuration mode.
Step 11	Router# show atm vc	Verify the ATM PVC configuration.

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Note When verifying your ATM PVC connectivity, note that you cannot issue the ping command over a voice PVC because the command applies to data only. If you have data and voice PVCs set to the same destination, you can issue the ping command over the data PVC.

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Preparing to Configure Voice Dial Peers

After you have analyzed your dial plan and decided how to integrate it into your existing network, you are ready to configure your network devices to support Voice over ATM. The actual configuration procedure depends on the topology of your voice network.

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TimeSaver If possible, you might want to configure the ATM dial peers in a back-to-back configuration before separating them across the ATM network. Using a back-to-back configuration, you can test your Voice over ATM and dial-peer configuration to see if you can successfully make a voice connection. Then, when you place both peers on the network, if you cannot make a voice connection, you can isolate the cause as a network problem.

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Creating a Peer Configuration Table

There is specific information relative to each dial peer that you must identify before you can configure Voice over ATM. One way to do this is to create a peer configuration table.

Figure 1 shows a diagram of a small voice network in which Router 1, with ATM virtual circuit 20, connects a small sales branch office to the main office through Router 2. There are only two devices in the sales branch office that need to be established as dial peers: a basic telephone and a fax machine. Router 2, with an ATM virtual circuit of 40, is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. There are three basic telephones connected to the PBX that need to be established as dial peers in the main office.

Table 1 shows the peer configuration table for the example illustrated in Figure 1.

Table 1: Peer Configuration Table for Sample Voice over ATM Network

Dial Peer	Extension	Prefix	Dest-Pattern	Type	Voice Port	Session Target
Router 1
1	61111		+13101161111	POTS	2/0/0
2	62222		+13101162222	POTS	2/0/1
10			+1310117....	VoATM		S2 20
Router 2
11			+1310116....	VoATM		S2 40
3	73333	7	+1310117....	POTS	2/1/0
4	74444	7	+1310117....	POTS	2/1/0
5	75555	7	+1310117....	POTS	2/1/0

The dial plan shown in lists a simple dial-peer configuration table with no special configuration for how you forward or playout excess digits. For more information on other options for designing your dial plan and configuring your dial peers to connect with PBXs, see the "Advanced Dial Peer Configuration" section in the "Configuring Voice over Frame Relay" chapter in the Multiservice Applications Configuration Guide for the Cisco IOS Release 12.1. The concepts described in that section also apply to Voice over ATM.

Configuring Dial Peers

Dial peers describe the entities to and from which a call is established. Dial-peer configuration tasks define the address or set of addresses serviced by that dial peer and the call parameters required to establish a call to and from that dial peer.

• POTS---Dial peer connected by a traditional telephony network. POTS dial peers point to a particular voice port on a voice-network device.

• Voice over ATM---Dial peer connected by an ATM WAN backbone. Voice over ATM dial peers point to specific voice-network devices.

Establishing two-way communication by using Voice over ATM requires establishing a specific voice connection between two defined endpoints. As shown in Figure 2, for outgoing calls (from the perspective of the POTS dial peer 1), the POTS dial peer establishes the source (the originating telephone number and voice port) of the call. The Voice over ATM dial peer establishes the destination by associating the destination phone number with a specific ATM virtual circuit.

To complete the two-way communications loop, configure Voice over ATM dial peer 2 as shown in Figure 3.

The only exception is when both POTS dial peers are connected to the same router, as shown in Figure 4. In this circumstance, because both dial peers share the same destination IP address, you do not need to configure a Voice over ATM dial peer.

When configuring dial peers, you need to understand the relationship between the destination pattern and the session target. The destination pattern represents the pattern for the device at the voice connection endpoint, such as a telephone or a PBX. The session target represents the serial port on the peer router at the other end of the ATM connection. Figure 5 and Figure 6 show the relationship between the destination pattern and the session target, as seen from the perspective of both routers in a Voice over ATM configuration.

Configuring POTS Dial Peers

To configure a POTS dial peer, identify the peer (by assigning it a unique tag number), define its telephone number, and associate it with a voice port through which calls can be established. Under most circumstances, the default values for the remaining dial-peer configuration commands are sufficient to establish connections.

Configuring Voice over ATM Dial Peers

To configure a Voice over ATM dial peer, you need to uniquely identify the peer by assigning it a unique tag number and define the outgoing serial port number and the virtual circuit number.

Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router

To configure Cisco-trunk permanent calls on a Cisco 3600 series router for Voice over ATM, use the following commands from interface configuration mode:

	Command	Purpose
Step 1	For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group	Identify the voice port you want to configure and enter voice-port configuration mode.
Step 2	Router(config-voiceport)# connection trunk destination-string [answer-mode]	Configure the trunk connection and specify the telephone number in the destination-string. When configuring Cisco-trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port operates in master mode. Enter the answer-mode keyword to specify that the voice port operates in slave mode.
Step 3	Router(config-voiceport)# shutdown	Shut down the voice port.
Step 4	Router(config-voiceport)# no shutdown	Reactivate the voice port to enable the trunk connection to take effect.

Note Every time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

Verifying the Voice Connection

Verify that the voice connection is working by doing the following:

• Pick up the handset on a telephone connected to the configuration and verify that you can get a dial tone.

• Make a call from the local telephone to a configured dial peer and verify that the call attempt is successful.

You can check the validity of your dial-peer and voice-port configuration by performing the following tasks:

• If you have relatively few dial peers configured, you can use the show dial-peer voice command to verify that the data configured is correct.

• To show the status of the voice ports, use the show voice port command.

• To show the call status for all voice ports, use the show voice call command.

• To show the current status of all DSP voice channels, use the show voice dsp command.

Troubleshooting Tips

If you are having trouble connecting a call and you suspect the problem is associated with the dial-peer configuration, you can try to resolve the problem by performing the following tasks:

• Use the show dial-peer voice command on the local and remote concentrators to verify that the data is configured correctly on both.

• Use the show interface command to verify that the ATM interface is up.

• Make sure the voice port, serial port and/or controller T1 0 is set to no shutdown.

Voice over ATM Configuration Example

The following is a sample configuration for Voice over ATM on the Cisco 3600 series:

version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname c3640_1
!
!
!
no ip subnet-zero
no ip routing
ip wccp version 2
ip host keyer-ultra 171.71.20.62
!
!
!
!
dial-control-mib max-size 500
!
process-max-time 200
!
interface Ethernet0/0
 ip address 172.28.129.54 255.255.255.192
 ip helper-address 171.71.20.62
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
!
interface Serial0/0
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 no fair-queue
!
interface Ethernet0/1
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
!
interface ATM1/0
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 no atm ilmi-keepalive
 pvc 1/100
  vbr-rt 1000 500
  encapsulation aal5mux voice
 !
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/1
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 no atm ilmi-keepalive
 pvc 2/100
  vbr-rt 1000 500
  encapsulation aal5mux voice
 !
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/1.1 point-to-point
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 pvc 3/200
  vbr-rt 64 64 4
  encapsulation aal5mux voice
!
!
interface ATM1/2
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/3
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/4
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/5
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/6
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM1/7
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 shutdown
 no atm ilmi-keepalive
 no scrambling-payload
 impedance 120-ohm
 no fair-queue
!
interface ATM3/0
 no ip address
 no ip directed-broadcast
 no ip route-cache
 no ip mroute-cache
 map-group atm1
 atm clock INTERNAL
 pvc 2/200
  encapsulation aal5snap
 no atm auto-configuration
 no atm ilmi-keepalive
 no atm address-registration
 no atm ilmi-enable
 pvc voice 1/100
  vbr-rt 5000 2500
  encapsulation aal5mux voice
!
!
ip default-gateway 172.28.129.1
ip classless
ip route 171.70.20.62 255.255.255.255 172.28.129.1
no ip http server
!
!
map-list atm1
 ip 4.4.4.2 atm-vc 2 broadcast
!
map-class frame-relay fr1
!
map-class frame-relay voice
 no frame-relay adaptive-shaping
 frame-relay cir 128000
 frame-relay bc 128000
snmp-server engineID local 00000009020000107BC778C0
snmp-server community public RO
snmp-server community SNMPv2c view v2default RO
snmp-server community v2 view v1default RO
snmp-server community config view v1default RO
snmp-server community voice view v1default RO
snmp-server packetsize 4096
snmp-server enable traps snmp
snmp-server enable traps casa
snmp-server enable traps config
snmp-server enable traps voice poor-qov
snmp-server host 171.71.128.229 version 2c SNMPv2c  config voice snmp
snmp-server host 171.71.128.242 version 2c public config voice snmp
snmp-server host 171.71.129.16 version 2c public tty frame-relay isdn hsrp
config entity envmon bgp rsvp rtr syslog stun sdllc dspu rsrb dlsw sdlc snmp
snmp-server host 171.71.129.164 version 2c public  config voice snmp
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 session-timeout 10
 password apple
 login
!
voice-port 2/0/0
 input gain 5
 output attenuation 5
!
voice-port 2/0/1
 input gain 5
 output attenuation 5
!
voice-port 2/1/0
 input gain 5
 output attenuation 5
!
voice-port 2/1/1
 input gain 5
 output attenuation 5
!
dial-peer voice 2 pots
 destination-pattern 4001
!
dial-peer voice 8000 pots
 destination-pattern 84000
!
dial-peer voice 9000 pots
 destination-pattern 94000
!
dial-peer voice 9001 pots
 destination-pattern 94001
!
dial-peer voice 348 voatm
 destination-pattern 348....
 signal-type ext-signal
 session target ATM3/0 pvc 1/100
!
dial-peer voice 338 voatm
 destination-pattern 338....
 signal-type ext-signal
 session target ATM1/0 pvc 1/100
!
dial-peer voice 2222 voatm
 preference 1
 session target ATM1/0 pvc 1/100
!
dial-peer voice 9500 voatm
 destination-pattern 95...
 session target ATM3/0 pvc 1/100
!
dial-peer voice 8400 pots
 destination-pattern 84000
!
dial-peer voice 50000 voatm
 destination-pattern 5264000
 session target ATM3/0 pvc 1/100
!
dial-peer voice 10000 pots
 destination-pattern 5254000
 port 2/0/0
!
dial-peer voice 10001 pots
 destination-pattern 4000789
 port 2/1/0
!
num-exp 1 1234
num-exp 2 2234
num-exp 12 34567890
num-exp 55 66666
end

Command Reference

This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command reference publications.

New and modified debug commands can be found in the "Debug Commands" section.

The following new and modified commands are described in this section (modified commands are marked by an asterisk):

• dial-peer hunt

• dial-peer terminator*

• dial-peer voice*

• encapsulation*

• forward-digits*

• huntstop*

• session target*

• signal-type*

• vbr-rt*

• voice hunt user-busy

dial-peer hunt

To specify a hunt selection order for dial-peers, use the dial-peer hunt dial-peer configuration command. Use the no form of this command to restore the default selection order.

Syntax Description

hunt-order-number

A number from 0 to 7 that selects a predefined hunting selection order: 0---Longest match in phone number, explicit preference, random selection. This is the default hunt order number. 1---Longest match in phone number, explicit preference, least recent use. 2---Explicit preference, longest match in phone number, random selection. 3---Explicit preference, longest match in phone number, least recent use. 4---Least recent use, longest match in phone number, explicit preference. 5---Least recent use, explicit preference, longest match in phone number. 6---Random selection. 7---Least recent use.

Defaults

The default is the longest match in phone number, explicit preference, and random selection (hunt order number 0).

Command Modes

Global configuration

Command History

Release	Modification
12.0(7)XK	This command was introduced and supported on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice access concentrator.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use the dial-peer hunt dial-peer configuration command if you have configured hunt groups. "Longest match in phone number" refers to the destination pattern that matches the greatest number of the dialed digits. "Explicit preference" refers to the preference setting in the dial-peer configuration. "Least recent use" refers to the destination pattern that has waited the longest since being selected. "Random selection" weight all of the destination patterns equally in a random selection mode.

Examples

The following example displays configuring the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection.

router# configure terminal
router(config)# dial-peer hunt 0

Related Commands

Command	Description
preference	Specifies the preferred selection order of a dial peer within a hunt group.
destination-pattern	Specifies the prefix or the complete telephone number for a dial peer.
show dial-peer voice	Displays configuration information for dial peers.

dial-peer terminator

To change the character used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to restore the default terminating character.

Syntax Description

character

Designates the terminating character for a variable-length dialed number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, a, b, c, and d. The default is #.

Defaults

The default terminating character is #.

Command Modes

Global configuration

Command History

Release	Modification
12.0	This command was introduced.
12.0(7)XK	Usage was restricted to variable-length dialed numbers.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

There are certain areas in the world (for example, in certain European countries) where telephone numbers can vary in length. When a dialed-number string has been identified as a variable length dialed-number, the system does not place a call until the configured value for the timeouts interdigits command has expired, or until the caller dials the terminating character. Use the dial-peer terminator global configuration command to change the terminating character.

Examples

The following example displays specifying "9" as the terminating character for variable-length dialed numbers:

router# configure terminal
router(config)# dial-peer terminator 9#

Related Commands

Command	Description
answer-address	Specifies the preferred selection order of a dial peer within a hunt group.
destination-pattern	Specifies the prefix or the complete telephone number for a dial peer.
timeouts interdigit	Specifies the interdigit timeout value for a voice port in seconds.
show dial-peer voice	Displays configuration information for dial peers.

dial-peer voice

To enter dial-peer configuration mode and specify the method of voice encapsulation, use the dial-peer voice global configuration command.

For the Cisco 2600 series:

For the Cisco 3600 series and the Cisco MC3810:

Syntax Description

tag	A number identifying a particular dial peer. Valid entries are 1 to 2147483647.
pots	POTS dial peer using basic telephone service.
voip	VoIP dial peer using voice encapsulation on the POTS network.
vofr	Voice over Frame Relay dial peer using encapsulation on the Frame Relay backbone network.
voatm	(Cisco 3600 and MC3810 only) Voice over ATM dial peer using real-time AAL5 voice encapsulation on the ATM backbone network.

Defaults

No default behavior or values.

Command Modes

Global configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.
11.3(1)MA	This command was first supported on the Cisco MC3810 with support for POTS, VoFR and VoATM.
12.0(3)XG	This command added VoFR to the Cisco 2600 and Cisco 3600 series routers.
12.0(4)T	This command added VoFR to the Cisco 7200 series platform.
12.0(7)XK	This command added VoIP to the Cisco MC3810 and VoATM to the Cisco 3600 series routers. Support for VoHDLC on the Cisco MC3810 was removed in this release.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use the dial-peer voice global configuration command to switch to the dial-peer configuration mode from the global configuration mode. Use the exit command to exit the dial-peer configuration mode and return to the global configuration mode.

Examples

The following example displays accessing dial-peer configuration mode and configuring a POTS peer identified as dial peer 10:

router# configure terminal
router(config)# dial-peer voice 10 pots
 

The following example displays accessing dial-peer configuration mode and configuring a VoATM peer identified as dial peer 20:

router# configure terminal
router(config)# dial-peer voice 20 voatm

Related Commands

Command	Description
voice-port	Enters voice-port configuration mode.

encapsulation

To configure the ATM adaptation layer (AAL) and encapsulation type for an ATM PVC class, use the encapsulation command in the appropriate command mode. Use the no form of this command to remove an encapsulation from a PVC class.

encapsulation aal-encap

no encapsulation aal-encap

---

Note This document only describes encapsulation settings for Voice over ATM. For the full syntax of the encapsulation command, refer to Cisco IOS 12.1 Wide Area Networking Command Reference.

---

Syntax Description

aal-encap

ATM adaptation layer (AAL) and encapsulation type. Possible values for aal-encap are as follows: aal5mux voice---For a MUX-type virtual circuit for Voice over ATM. aal5snap---The only encapsulation supported for Inverse ARP. Logical Link Control/Subnetwork Access Protocol (LLC/SNAP) precedes the protocol datagram.

Defaults

The global default encapsulation is aal5snap. See the "Usage Guidelines" section for other default characteristics.

Command Modes

Interface-ATM-VC configuration (for an ATM PVC or SVC)

Command History

Release	Modification
11.3 T	This command was introduced.
12.0	This command superseded the encapsulation atm command for the Cisco MC3810, and the aal5mux frame and aal5mux voice suboptions appeared.
12.0(7)XK	Support for the aal5mux voice option was added to the Cisco 3600 series routers.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use one of the aal5mux encapsulation options to dedicate the specified PVC to a single protocol; use the aal5snap encapsulation option to multiplex two or more protocols over the same PVC. Whether you select aal5mux or aal5snap encapsulation depends on practical considerations, such as the type of network and the pricing offered by the network. If the network's pricing depends on the number of PVCs set up, aal5snap may be the appropriate choice. If pricing depends on the number of bytes transmitted, aal5mux may be the appropriate choice because it has slightly less overhead.

If you specify virtual template parameters after the ATM PVC is configured, issue a shutdown command followed by a no shutdown command on the ATM subinterface to restart the interface, causing the newly configured parameters (such as an IP address) to take effect.

Examples

The following example displays configuring a PVC to support encapsulation for Voice over ATM:

router(config-if)# pvc 20
router(config-if-atm-pvc)# encapsulation aal5mux voice

forward-digits

To specify which digits to forward for voice calls, use the forward-digits dial-peer configuration command. If the no form of this command is entered, any digits not matching the destination-pattern are not forwarded. Use the default form of this command to restore the default state.

Syntax Description

num-digit	The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. The valid range is 0 to 32. Setting the value to 0 is equivalent to entering no forward-digits.
all	Forward all digits. If all is entered, the full length of the destination pattern is used.
extra	If the length of the dialed digit string is greater than the length of the dial-peer destination pattern, the extra right-justified digits are forwarded. However, if the dial-peer destination pattern is variable length (ending with character "T", for example: T, 123T, 123...T), extra digits are not forwarded.

Defaults

Dialed digits not matching the destination-pattern are forwarded.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1) MA	This command was introduced on the Cisco MC3810.
12.0(2)T	The implicit option was added.
12.0(4)T	This command was modified to support ISDN PRI QSIG signaling calls.
12.0(7)XK	This command was first supported on the Cisco 2600 and 3600 platforms, the implicit keyword was removed, and the extra keyword was added.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

This command applies only to POTS dial peers.

Forwarded digits are always right-justified, so that extra leading digits are stripped.

The destination pattern includes both explicit digits and wildcards if present.

Use the default form of this command if a non-default digit-forwarding scheme was entered previously and you want to restore the default.

For QSIG ISDN connections, entering forward-digits all implies that all of the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.

Examples

The following example displays forwarding all of the digits in the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 8...
router(config-dial-peer)# forward-digits all
 

The following example displays forwarding 4 of the digits in the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 555....
router(config-dial-peer)# forward-digits 4
 

The following example displays forwarding the extra right-justified digits that exceed the length of the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 555....
router(config-dial-peer)# forward-digits extra

Related Commands

Command	Description
destination-pattern	Defines the prefix or the full E.164 telephone number to be used for a dial peer.
show dial-peer voice	Displays configuration information for dial peers.

huntstop

To disable all further dial-peer hunting if a call fails when using hunt groups, enter the huntstop dial-peer configuration command. To reenable dial peer call hunting, enter the no form of this command.

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Modes

Dial-peer configuration

Command History

Release	Modification
12.0(5)T	This command was introduced on the Cisco MC3810.
12.0(7)XK	Support for this command was extended to the Cisco 2600 and 3600 series routers.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Once you enter this command, no further hunting is allowed if a call fails on the specified dial peer.

This command can be used with all types of dial peers.

Examples

The following example shows how to disable dial-peer hunting on a specific dial peer:

router(config)# dial peer voice 100 voatm
router(config-dial-peer)# huntstop
 

The following example shows how to reenable dial-peer hunting on a specific dial peer:

router(config)# dial peer voice 100 voatm
router(config-dial-peer)# no huntstop

Related Commands

Command	Description
dial-peer voice	Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

session target

To configure a network-specific address for a dial peer, use the session target dial-peer configuration command. Use the no form of this command to disable this feature.

For Cisco 3600 Voice over ATM dial peers:

Syntax Description

interface	Interface type and interface number on the router.
pvc	The specific ATM permanent virtual circuit (PVC) for this dial peer.
name	The PVC name.
vpi/vci	ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC. On the Cisco 3600, if you have the Multiport T1/E1 ATM network module with IMA installed, the valid range for vpi is 0-15, and the valid range for vci is 1-255. If you have the OC3 ATM Network Module installed, the valid range for vpi is 0-15, and the valid range for vci is 1-1023.
vci	ATM network virtual channel identifier (VCI) of this PVC.

Defaults

Enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.
11.3(1)MA	Support was added for VoFR, VoATM, and VoHDLC dial peers on the Cisco MC38110.
12.0(3) XG	The cid option was added. Support was added for VoFR dial peers on the Cisco 2600 and Cisco 3600 series routers.
12.0(4)T	This command was implemented in Cisco IOS Release 12.0(4)T.
12.0(7)XK	Support was added for VoATM dial peers on the Cisco 3600 series routers. Support was added for VoIP dial peers on the Cisco MC3810. Support for VoHDLC on the Cisco MC3810 was removed in this release.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.

Examples

The following example displays configuring a session target for Voice over ATM on the Cisco 3600 series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.

router(config)# dial-peer voice 12 voatm
router(config-dial-peer)# destination-pattern 13102221111
router(config-dial-peer)# session target atm1/0 pvc 1/100

Related Commands

Command	Description
called-number	Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
codec (dial-peer)	Specifies the voice coder rate of speech for a dial peer.
cptone	Specifies a regional tone, ring, and cadence setting for an analog voice port.
destination-pattern	Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.
dtmf-relay	Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference	Indicates the preferred selection order of a dial peer within a hunt group.
session protocol	Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
signal-type	Sets the signaling type to be used when connecting to a dial peer.

signal-type

To set the signaling type to be used when connecting to a dial peer, use the signal-type command from dial-peer configuration mode. To return to the default signal-type, use the no form of this command.

Syntax Description

cas	North American EIA-464 Channel-Associated Signaling (robbed bit signaling). If the Digital T1 Packet Voice Trunk Network Module is installed, this option may not be available.
cept	Provides a basic E1 ABCD signaling protocol. Used primarily for E&M interfaces. When used with FXS/FXO interfaces, this protocol is equivalent to MELCAS.
ext-signal	External signaling. The DSP does not generate any signaling frames. Use this option when there is an external signaling channel, for example, CCS, or when you need to have a permanent "dumb" voice pipe.
transparent	On the Cisco MC3810, selecting this option produces different results depending on whether you are using a digital voice module (DVM) or an analog voice module (AVM). For a DVM: The ABCD signaling bits are copied from or transported through the T1/E1 interface "transparently" without modification or interpretation. This enables the MC3810 to handle arbitrary or unknown signaling protocols. For an AVM: It is not possible to provide "transparent" behavior because the Cisco MC3810 must interpret the signaling information in order to read and write the correct state to the analog hardware. This option is mapped to be equal to "cas."

Defaults

cas

Command Modes

Dial-peer configuration

Command History

Release	Modification
12.0(3)XG	This command was introduced.
12.0(4)T	Support was added for the Cisco 7200 series routers.
12.0(7)XK	In previous releases, the cept and transparent options were only supported on the Cisco MC3810. Beginning in this release, these options are supported on the Cisco 2600, Cisco 3600 and Cisco 7200 routers.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

This command applies to VoFR and VoATM dial peers. It is used with permanent connections only (Cisco trunks and FRF.11 trunks), not with switched calls.

This command is used to inform the local telephony interface of the type of signaling it should expect to receive from the far-end dial peer. To turn signaling off at this dial peer, select the ext-signal option. If signaling is turned off and there are no external signaling channels, a "hot" line exists, enabling this dial peer to connect to anything at the far end.

When you connect an FXS to another FXS, or if you have anything other than an FXS/FXO or E&M/E&M pair, the appropriate signaling type on Cisco 2600 series and 3600 series routers is ext-signal (disabled).

If you have a digital E1 connection at the remote end that is running cept/MELCAS signaling and you then trunk that across to an analog port, you should make sure that you configure both ends for the cept signal-type.

If you have a T1 or E1 connection at both ends and the T1/E1 is running a signaling protocol that is neither EIA-464 or cept/MELCAS, you may want to configure the signal-type for the transparent option in order to pass through the signaling.

Examples

The following example shows how to disable signaling on a Cisco 2600 series or 3600 series router or on an MC3810 concentrator for VoFR dial peer 200, starting from global configuration mode:

router(config)# dial-peer voice 200 vofr
router(config-dial-peer)# signal-type ext-signal
router(config-dial-peer)# 

Related Commands

Command	Description
codec (dial-peer)	Specifies the voice coder rate of speech for a dial peer.
connection	Specifies the connection mode for a voice port.
destination-pattern	Specifies the telephone number associated with a dial peer.
dtmf-relay	Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
preference	Enables the preferred dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string.
session protocol	Establishes the VoFR protocol for calls between local and remote routers.
session target	Specifies a network-specific address for a dial peer.
sequence-numbers	Enables the generation of sequence numbers in each frame generated by the DSP.

vbr-rt

To configure the real-time variable bit rate (VBR) for Voice over ATM connections, use the vbr-rt ATM virtual circuit configuration command. Use the no form of this command to restore the default.

vbr-rt peak-rate average-rate burst

no vbr-rt

Syntax Description

peak-rate	The peak information rate (PIR) of the voice connection in kbps. The range is 56 to 10000.
average-rate	The average information rate (AIR) of the voice connection in kbps. The range is 1 to 56.
burst	Burst size in number of cells. The range is 0 to 65536.

Defaults

No vbr-rt settings are configured.

Command Modes

ATM virtual circuit configuration

Command History

Release	Modification
12.0	This command was introduced on the Cisco MC3810.
12.0(7)XK	Support for this command was extended to the Cisco 3600 series.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

The vbr-rt command configures traffic shaping between voice and data PVCs. Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will be carrying bursty traffic. The peak, average, and burst values are needed so the PVC can effectively handle the bandwidth for the number of voice calls. To calculate the minimum peak, average, and burst values for the number of voice calls, use the following calculations:

• Peak value: (2 x the maximum number of calls) x 16 kb

• Average value: (1 x the maximum number of calls) x 16 kb

• Burst value: (4 x the maximum number of calls)

---

Note When you configure data PVCs that will be traffic shaped with voice PVCs, use the aalsnap encapsulation and calculate the overhead as 1.13 times the voice rate.

---

Examples

The following example displays configuring the traffic shaping rate for ATM PVC 20 on a Cisco 3600. In the example, the peak, average and burst rates are calculated based on a maximum of 20 calls on the PVC.

router(config-if)# pvc 20
router(config-if-atm-pvc)# encapsulation aal5mux voice
router(config-if-atm-pvc)# vbr-rt 640 320 80

Related Commands

Command	Description
encapsulation	Configures the ATM adaptation layer (AAL) and encapsulation type for an ATM PVC class

voice hunt user-busy

To configure an originating or tandem router so it continues dial-peer hunting if it receives a user-busy disconnect code from a destination router, use the voice hunt user-busy command in global configuration mode. To configure the router so it stops dial-peer hunting if it receives a user-busy disconnect code (the default option), use the no form of this command.

Router(config)# voice hunt user-busy

Debug Commands

This section provides information on new and modified VoATM debug commands for the Cisco 3600 series.

All other debug commands used with Voice over ATM are documented in the Cisco IOS Debug Command Reference for Cisco IOS Release 12.1.

The following new and modified commands are described in this section:

• debug ccswvoice voatm-debug

• debug ccswvoice voatm-session

debug ccswvoice voatm-debug

To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice vooatm-debug command from privileged EXEC mode. Use the no form of this command to turn off the debug function.

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)MA	This command was introduced on the Cisco MC3810.
12.0(7)XK	This command was first supported on the Cisco 3600 series.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use this command when attempting to troubleshoot a VoATM call that uses the "cisco-switched" session protocol.This command provides the same information as the debug ccswvoice voatm-session command, but includes additional debugging information relating to the calls.

Examples

The following example shows sample output from the debug ccswvoice voatm-debug command:

router# debug ccswvoice voatm-debug
 
2w2d: ccswvoice: callID 529927 pvcid -1 cid -1 state NULL event O/G SETUP
2w2d: ccswvoice_out_callinit_setup: callID 529927 using pvcid 1 cid 15
2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G INIT event I/C PROC
2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G PROC event I/C ALERTccfrf11_caps_ind: codec(preferred) = 1
 
2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G ALERT event I/C CONN
2w2d: ccswvoice_bridge_drop: dropping bridge calls src 529927 dst 529926 pvcid 1 cid 15 state ACTIVE
2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state ACTIVE event O/G REL
2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state RELEASE event I/C RELCOMP
2w2d: ccswvoatm_store_call_history_entry: cause=10 tcause=10    cause_text=normal call clearing.

Related Commands

Command	Description
debug ccswvoice voatm-session	Displays the ccswvoice function calls during call setup and teardown.

debug ccswvoice voatm-session

To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice voatm-session command from privileged EXEC mode. Use the no form of this command to turn off the debug function.

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)MA	This command was introduced on the Cisco MC3810.
12.0(7)XK	This command was first supported on the Cisco 3600 series.
12.1(2)T	This command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use this command to show the state transitions of the cisco-switched-voatm state machine as a call is processed. This command should be used when attempting to troubleshoot a VoATM call that uses the "cisco-switched" session protocol.

Examples

The following example shows sample output from the debug ccswvoice voatm-session command:

router# debug ccswvoice voatm-session
 
2w2d: ccswvoice: callID 529919 pvcid -1 cid -1 state NULL event O/G SETUP
2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G INIT event I/C PROC
2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G PROC event I/C ALERT
2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G ALERT event I/C CONN
2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state ACTIVE event O/G REL
2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state RELEASE event I/C RELCOMP
 

Related Commands

Command	Description
debug ccswvoice voatm-debug	Displays the ccswvoice function calls during call setup and teardown.

Glossary

AAL---ATM Adaptation Layer.

Call leg---A logical connection between the router and either a telephony endpoint over a bearer channel or another endpoint using a session protocol.

CAS---Channel associated signaling.

Codec---Coder-decoder. 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 ATM, it specifies the voice coder rate of speech for a dial peer.

Dial peer---An addressable call endpoint. In Voice over ATM, there are two kinds of dial peers: POTS and VoATM.

DS0---A 64-K B-channel on an E1 or T1 WAN interface.

DTMF---Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).

E&M---E&M stands for recEive and transMit (or Ear and Mouth). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines).

FXO---Foreign Exchange Office. An FXO interface connects to the PSTN's central office and is the interface offered on a standard telephone. Cisco's FXO interface is an RJ-11 connector that allows an analog connection to be directed at the PSTN's central office. This interface is of value for off-premise extension applications.

FXS---Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.

PBX---Private Branch Exchange. Privately owned central switching office.

PLAR---Private Line Auto Ringdown. This type of service results in a call attempt to some particular remote endpoint when the local extension is taken off-key.

POTS---Plain Old Telephone Service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the public switched telephone network.

POTS dial peer---Dial peer connected via 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.

PVC---Permanent virtual circuit.

TDM---Time division multiplexing.

Trunk---Service that allows quasi-transparent connections between two PBXs, a PBX and a local extension, or some other combination of telephony interfaces to be permanently connected together by the session application and signaling passed transparently through the IP network.

VBR---Variable Bit Rate.

VoATM dial peer---Dial peer connected by an ATM network. VoATM peers point to specific VoATM devices.

Posted: Sun May 21 10:44:04 PDT 2000
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