Table Of Contents
Configuring T1/E1 High Capacity Digital Voice Port Adapters
VoIP: T1/E1 Timing, Signaling, Framing, and Line Encoding
Single T1/E1 Port Provides Clocking
Single T1/E1 Port Receiving Clock from the Line
Dual T1/E1s, Both Receive Clocking from the Line
Dual T1s, One Receives Clocking and One Provides Clocking
Dual T1s, Both Clocks from Router
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring the DSPfarm Interface
Configuring Card Type and T1 Controller Settings
Configuring Card Type and E1 Controller Settings
Verifying Card Type and Controller Settings
Verifying Serial Interface Configuration
Monitoring and Maintaining T1/E1 Digital Voice Configuration
Routed Digits — Switched VoIP Calls
Routing Calls Through an H.323 Gatekeeper
Private-Line Auto-Ringdown Configuration — Switched VoIP Calls
Connection Trunk Configuration — Permanent VoIP Calls
Drop-and-Insert Sample Configuration
Configuring T1/E1 High Capacity Digital Voice Port Adapters
This document describes how to configure T1/E1 high capacity digital voice port adapters on
Cisco 7200 series routers includes the following sections:•
Supported Standards, MIBs, and RFCs
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Feature Overview
T1/E1 high capacity digital voice port adapters for Cisco 7200 series routers allow enterprises or service providers, using the equipped routers as customer premises equipment, to deploy digital voice and fax relay. These port adapters receive constant bit-rate telephony information over T1 interfaces and can convert that information to a compressed format and be transmitted as voice over IP (VoIP).
Cisco IOS software can be configured to support a variety of applications, including:
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For more information about these applications, see "Configuration Examples" on page 35.
T1/E1 digital voice over IP includes the following functionality:
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Benefits
T1/E1 high capacity digital voice port adapters allow Cisco 7200 series routers to provide T1/E1 connectivity to PBXs or to a central office (CO). With digital T1/E1 connectivity, Cisco 7200 series routers can provide greater voice density for enterprise and service provider VoIP networks. A T1/E1 voice port adapter is a complete solution, made up of a port adapter with installed packet voice data modules, and a T1/E1 interface with two T1/E1 ports.
VoIP: T1/E1 Timing, Signaling, Framing, and Line Encoding
With the introduction of the T1/E1 high capacity digital voice port adapters for the Cisco 7200 series routers, you must set timing, signaling, framing, and line encoding. The T1/E1 high capacity digital voice port adapters can connect to either a PBX (or similar telephony device) or to a Central Office (CO) in order provide PSTN connectivity.
The differences that set T1/E1 digital configuration apart from analog configuration are as follows:
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Timing
This section describes the five basic timing scenarios that can occur when a digital T1/E1 voice port adapter is connected to a PBX, CO, or both. In all of the examples below, the PSTN (or central office) and the PBX are interchangeable for the purposes of providing or receiving clocking.
The digital T1/E1 port adapter has an on-board Phase-Lock Loop (PLL) chip that can either provide a clock source to both T1/E1s or receive clocking that can drive the second T1/E1. All timing commands are T1/E1 controller configuration commands.
Single T1/E1 Port Provides Clocking
In this scenario, the digital T1/E1 port adapter is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the T1/E1 line.
Figure 1 Single T1/E1 Port Providing Clock
The following configuration sets up this T1 clocking method:
controller T1 1/0framing esflinecoding b8zsclock source internalds0-group 1 timeslots 1-24 type e&m-winkThe following configuration sets up this E1 clocking method:
controller E1 1/0framing crc4linecoding hdb3clock source internalds0-group 1 timeslots 1-24 type e&m-wink
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Single T1/E1 Port Receiving Clock from the Line
In this scenario, the digital T1/E1 port adapter receives clocking from the connected device (CO or PBX). The PLL clocking is driven by the clock reference on the receiver (Rx) side of the T1/E1 connection.
Figure 2 Single T1 Receiving Clock from Line
The following configuration sets up this clocking method for the T1:
controller T1 1/0framing esflinecoding b8zsclock source lineds0-group 1 timeslots 1-24 type e&m-winkDual T1/E1s, Both Receive Clocking from the Line
In this scenario, the digital T1/E1 has two reference clocks, one from the PBX and another from the CO. Since the PLL can only derive clocking from one source, this case is more complex than the two preceding examples.
Before looking at the details, consider two important concepts that underlay the clocking method:
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Figure 3 Dual T1s Receiving Line Clocking
In this scenario, the PLL derives clocking from the CO and puts the T1 port connected to the PBX into looped-time mode. This is usually the best method because the CO provides an excellent clock source (and usually requires that it provide that source) and a PBX usually must receive clocking from the other T1.
The following configuration sets up this clocking method:
controller T1 1/0 << description - connected to the COframing esflinecoding b8zsclock source line primaryds0-group 1 timeslots 1-24 type e&m-wink!controller T1 1/1 << description - connected to the PBXframing esflinecoding b8zsclock source lineds0-group 1 timeslots 1-24 type e&m-winkThe clock source line primary command tells the router to use this T1 port to drive the PLL. All other T1 ports configured as clock source line are then put into an implicit loop-timed mode. If the primary T1 port fails or goes down, the other T1 instead receives the clock that drives the PLL. In this configuration, T1 1/1 may see controlled slips, but these should not force it down. This method prevents the PBX from seeing slips.
Dual T1s, One Receives Clocking and One Provides Clocking
In this scenario, the digital T1 port adapter receives clocking for the PLL from T1 0 and uses this clock as a reference to clock T1 1. If T1 0 fails, the PLL internally generates the clock reference to drive T1 1.
Figure 4 Dual T1s, One Receiving and One Providing Clocking
The following configuration sets up this clocking method:
controller T1 1/0framing esflinecoding b8zsclock source lineds0-group 1 timeslots 1-24 type e&m-wink!controller T1 1/1framing esflinecoding b8zsclock source internalds0-group 1 timeslots 1-24 type e&m-winkDual T1s, Both Clocks from Router
In this scenario, the router is "Master of the Timing Universe," generating the clock for the PLL and therefore for both T1s.
Figure 5 Dual T1s, Both Clocks from Router
The following configuration sets up this clocking method:
controller T1 1/0framing esflinecoding b8szclock source internalds0-group 1 timeslots 1-24 type e&m-wink!controller T1 1/1framing esflinecoding b8zsclock source internalds0-group 1 timeslots 1-24 type e&m-winkSignaling
There are three types of signaling that you should consider for digital T1/E1:
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controller T1 1/0ds0-group 1 timeslots 1-24 type e&m-wink-start
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controller T1 1/0ds0-group 1 timeslots 1-24 type fxo-ground-startor
controller T1 1/0ds0-group 1 timeslots 1-24 type fxs-loop-start
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Framing
T1 high capacity digital voice port adapters support two types of framing for T1 CAS: ESF (Extended Superframe) and SF (Super Frame), also called D4 framing. The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines T1 framing, as in the following example:
controller T1 1/0framing esfor
controller T1 1/0framing sfLine Encoding
T1 high capacity digital voice port adapters support two types of framing for T1 CAS: B8ZS (bipolar-8 zero substitution) and AMI (alternate mark inversion). The line encoding of the router and switch (CO or PBX) must match. The linecoding controller configuration command defines T1 framing, as in the following example:
controller T1 1/0linecoding b8zsor
controller T1 1/0linecoding amiVerifying Configuration
Use the show controller privileged EXEC command to verify the proper digital T1 configuration:
router# show controller T1 1/0T1 1/0 is up.Applique type is Channelized T1Cablelength is short 110Description: Digital T1 WICNo alarms detected.Framing is ESF, Line Code is B8ZS, Clock Source is Line Primary.Data in current interval (2 seconds elapsed):0 Line Code Violations, 0 Path Code Violations0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail SecsUse the show controller privileged EXEC command to verify the proper digital E1 configuration:
Getty#show controller e1 2/0E1 2/0 is up.Applique type is Channelized E1 - balancedNo alarms detected.Framing is CRC4, Line Code is AMI, Clock Source is Line.International Bit: 1, National Bits: 00000Active xconns: 0Data in current interval (259 seconds elapsed):263359 Line Code Violations, 177418 Path Code Violations0 Slip Secs, 1 Fr Loss Secs, 354 Line Err Secs, 6 Degraded Mins355 Errored Secs, 1 Bursty Err Secs, 1 Severely Err Secs, 0 Unavail SecsData in Interval 1:0 Line Code Violations, 0 Path Code Violations0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail SecsTotal Data (last 1 15 minute intervals):0 Line Code Violations, 0 Path Code Violations,0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail SecsRestrictions
The following restrictions apply to digital T1/E1 voice port adapter configuration:
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Related Features and Technologies
VoIP Quality of Service
This section explains the quality issues that you should consider when building Voice over IP (VoIP) networks and offers a few tips about configuring VoIP with the appropriate quality of service (QoS):
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Supported Platforms
This feature is supported on the following platforms:
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Supported Standards, MIBs, and RFCs
RFCs
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MIBs
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International Telecommunication Union (ITU-T) G-Series Codec Compression Specifications
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Prerequisites
Digital T1/E1 voice requires specific service, software, and hardware:
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The memory required may be greater than listed above for high-volume applications.
Support for T1/E1 high capacity digital voice port adapters is included in Plus feature sets. The IP Plus feature set requires 16 Mb of flash memory.
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Single-Port 30 Channel T1/E1 High-Density Voice Port Adapter (PA-VXC-2TE1)
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Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0 provide information about setting up voice networks.
Configuration Tasks
Perform the following tasks to configure a digital T1/E1 voice port adapter:
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Configuring the DSPfarm Interface
Follow the procedure below to configure a DSPfarm interface. To learn more about other commands that can be used by the DSP interface, refer to the Cisco IOS Release 12.0 configuration guides.
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Configures the DSP interface.
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Specify the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs.
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The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command. See Step 7 in "Configuring Voice Dial Peers" on page 24.
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Add any additional configuration subcommands required to enable routing protocols and set the interface characteristics for your configuration requirements.
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Change the shutdown state to up and enable the interface.
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When you have included all the configuration subcommands to complete the configuration, press Ctrl-Z to exit configuration mode
After configuring the new DSPfarm interface, use show commands to display the status of the new interface, or to verify changes you have made.
For more information on how to configure the DSP interface, refer to the following publications:
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(Cisco IOS Release 12.0)Configuring Card Type and T1 Controller Settings
The following steps specify codec settings for card types and set up T1 controllers for clocking and other T1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for Drop-and-Insert capability.
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Enter global configuration mode.
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Enter T1 card type and specify the slot location by using a value from 0 to 5, depending upon your router.
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Router(config)# controller T1 slot/portEnter controller configuration mode for the T1 controller at the specified slot/port location. Valid values for slot and port are 0 and 1.
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Configure controller T1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the T1 controller ports:
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Set the framing according to your service provider's instructions. Choose Extended Superframe (ESF) format or Super Frame (SF) format.
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Set the line encoding according to your service provider's instructions. Bipolar-8 zero substitution (B8ZS) encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations. Alternate mark inversion (AMI) represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream.
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Router(config-controller)# cablelength long {-15db | -22.5db | -7.5db | 0db}or
cablelength short {110ft | 220ft | 330ft | 440ft | 550ft | 600ft}(T1/E1 interfaces only) The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected.
To set a cable length longer than 600 feet for a T1 link, use the cablelength long command. The keywords are as follows:
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To set a cable length 600 feet or less for a T1 link, use the cablelength short command. There is no default for cablelength short. The keywords are as follows:
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If you do not set the cable length, the system defaults to a setting of cablelength long 0db.
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This command defines the T1 channels for use by compressed voice calls as well as the signaling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration, as shown in Step 2. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.
ds0-group-no is a value from 0 to 23 that identifies the DS0 group.
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timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 timeslots, define additional groups. The system maps additional voice ports for each defined group. See Step 2 of "Configuring Voice Ports" on page 22.
The signaling method selection for type depends on the connection that you are making:
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(Optional) Use this command only when you need TDM channel groups for the Drop-and-Insert (also called TDM Cross-Connect) function with a two-port T1 trunk multiflex interface card.
tdm-group-no is a value from 1 to 31 that identifies the channel group.
timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. For T1, allowable values are from 1 to 24.
The signaling method selection for type depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line. Choose a type based on the criteria described above in Step 8.
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Activate the controller.
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Exit controller configuration mode. Skip the next step if you are not setting up the Drop-and-Insert capability.
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(Optional) This global configuration command sets up the connection between two T1 TDM groups of timeslots on the trunk interfaces—for the Drop-and-Insert capability.
id is a name for the connection.
Identify each T1/E1 controller by its slot/port location. Valid values for slot and port are 0 and 1.
tdm-group-no-1 and tdm-group-no-2 identify the TDM group numbers (from 1 to 31) on the specified controller. The groups were set up in Step 9.
See the "Configuration Examples" section for sample Drop-and-Insert configurations.
Repeat Steps 2 and 2 for each card type.
Repeat Steps 3 through 11 for each controller.
Configuring Card Type and E1 Controller Settings
The following steps specify codec settings for card types and set up E1 controllers for clocking and other E1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for Drop-and-Insert capability.
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Enter global configuration mode.
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Enter E1 card type and specify the slot location by using a value from 0 to 5, depending upon your router.
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Specify the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs.
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The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command. See Step 7 in "Configuring Voice Dial Peers" on page 24.
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Router(config)# controller E1 slot/portEnter controller configuration mode for the E1 controller at the specified slot/port location. Valid values for slot and port are 0 and 1.
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Configure controller E1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the E1 controller ports:
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Set the framing according to your service provider's instructions. Choose CRC4 format or NO CRC4 format.
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Set the line encoding according to your service provider's instructions.
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This command defines the E1 channels for use by compressed voice calls as well as the signaling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration, as shown in Step 2. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.
ds0-group-no is a value from 0 to 23 that identifies the DS0 group.
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timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 timeslots, define additional groups. The system maps additional voice ports for each defined group. See Step 2 of "Configuring Voice Ports" on page 22.
The signaling method selection for type depends on the connection that you are making:
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(Optional) Use this command only when you need TDM channel groups for the Drop-and-Insert (also called TDM Cross-Connect) function with a two-port T1/E1 multiflex trunk interface card.
tdm-group-no is a value from 1 to 31 that identifies the channel group.
timeslot-list is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of timeslots. For T1 or E1, allowable values are from 1 to 24.
The signaling method selection for type depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line. Choose a type based on the criteria described above in Step 8.
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Activate the controller.
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Exit controller configuration mode. Skip the next step if you are not setting up the Drop-and-Insert capability.
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(Optional) This global configuration command sets up the connection between two T1/E1 TDM groups of timeslots on the trunk interfaces—for the Drop-and-Insert capability.
id is a name for the connection.
Identify each E1 controller by its slot/port location. Valid values for slot and port are 0 and 1.
tdm-group-no-1 and tdm-group-no-2 identify the TDM group numbers (from 1 to 31) on the specified controller. The groups were set up in Step 9.
See the "Configuration Examples" section for sample Drop-and-Insert configurations.
Repeat Steps 2 and 2 for each card type.
Repeat Steps 3 through 11 for each controller.
Verifying Card Type and Controller Settings
To verify the configuration of card type and controller settings, follow these steps:
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Router# show running-config...hostname router-alphacard type t1 1dspint DSPfarm1/0codec high...Step 2
Router# show controller T1 1/0T1 1/0 is up.Applique type is Channelized T1Cablelength is short 110Description: T1 WIC card AlphaNo alarms detected.Framing is ESF, Line Code is B8ZS, Clock Source is Line Primary.Data in current interval (1 seconds elapsed):0 Line Code Violations, 0 Path Code Violations0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail SecsConfiguring Serial Interfaces
The way you set up serial and LAN interfaces depends on your application. To configure VoIP, you must at least set up IP addresses for serial interfaces. When a user dials enough digits to match a configured destination pattern, the telephone number is mapped to an IP host through the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a PBX that completes the call to the configured destination pattern.
This document does not explain all possible serial interface configuration options, nor does it show LAN interface configuration. For complete information, see the Cisco IOS Release 12.0 Cisco IOS Interface Configuration Guide and the Cisco IOS Interface Command Reference.
The "Configuration Examples" section shows a sample configuration that sets up VoIP over Frame Relay. For more information about setting up voice networks, see Voice, Video, and Home Applications Configuration Guide for Cisco IOS Release 12.0.
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Verifying Serial Interface Configuration
To verify the serial interface configuration, enter the privileged EXEC command show interface serial, which displays the status of all serial interfaces or of a specific serial interface, as shown in the following example. You can use this command to check the encapsulation, IP addressing, and other settings:
Router# show interface serial0/0:0Serial0/0:0 is up, line protocol is upHardware is QUICC SerialInternet address is 1.156.1.1/24MTU 1500 bytes, BW 1536 Kbit, DLY 20000 usec,reliability 255/255, txload 1/255, rxload 1/255Encapsulation HDLC, loopback not setKeepalive not setLast input 00:00:00, output 00:00:00, output hang neverLast clearing of "show interface" counters neverInput queue: 0/75/0 (size/max/drops); Total output drops: 0Queueing strategy: weighted fairOutput queue: 0/1000/64/0 (size/max total/threshold/drops)Conversations 0/1/256 (active/max active/max total)Reserved Conversations 0/0 (allocated/max allocated)5 minute input rate 1000 bits/sec, 1 packets/sec5 minute output rate 1000 bits/sec, 1 packets/sec637 packets input, 64736 bytes, 0 no bufferReceived 181 broadcasts, 0 runts, 5 giants, 0 throttles3617 input errors, 1506 CRC, 1646 frame, 0 overrun, 0 ignored, 0 abort682 packets output, 67213 bytes, 0 underruns0 output errors, 0 collisions, 1070 interface resets0 output buffer failures, 0 output buffers swapped out13 carrier transitionsTimeslot(s) Used:1-24, Transmitter delay is 0 flagsConfiguring Voice Ports
Follow these steps to set up voice ports to support the local and remote stations. Not all possible commands are shown here. To learn more, see Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
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Enter global configuration mode.
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Enter voice-port configuration mode.
slot is the router location where the voice port adapter is installed. Valid entries are from 0 to 3.
port indicates the voice interface card location. Valid entries are 0 or 1.
Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card. For more information about DS0 groups, see Step 11 of "Configuring the DSPfarm Interface" on page 12.
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(Optional) This command allows you to specify a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port.
For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.
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(Optional) This parameter is enabled by default. It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the resulting silence can fool the caller into thinking the call is disconnected instead of being merely idle.
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(Optional) This setting is enabled by default. Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25 milliseconds can cause problems.
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(Optional) This command adjusts the echo canceller by the specified number of milliseconds; the default is 16.
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(Optional) This command sets up a connection mode for the voice port.
plar specifies a Private Line Autoringdown (PLAR) connection, which rings a remote telephone when the dial peer goes off hook.
trunk specifies a straight tie-line connection to a PBX.
string specifies the remote telephone number or significant start digits of the number.
See the "Configuration Examples" section for sample PLAR and trunk configurations.
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(Optional) This command sets the number of seconds the system waits—after the caller has input the initial digit—for a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. The default value is 10 seconds, and the timeout can be set from 0 to 120 seconds.
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Exit voice-port configuration mode.
Verifying Voice Ports
Follow the procedure below to verify voice-port configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
Important command output is shown in bold.
To verify the voice-port configuration, enter the privileged EXEC show voice port slot/port:ds0-group command. The following sample output from the command shows explanatory information after the "<<" characters:
cisco-router# show voice port 1/0:1receEive and transMit Slot is 1, Sub-unit is 0, Port is 1 << voice-port 1/0:1Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for USConfiguring Voice Dial Peers
Follow these steps to set up voice dial peers to support the local and remote stations. Not all possible commands are shown here. To learn more, see Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
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Enter global configuration mode.
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Enter dial-peer configuration mode and define a local dial peer that will connect to the plain old telephone service (POTS) network.
number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647.
pots indicates a peer using basic telephone service.
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Configure the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number.
string is a series of digits that specify the E.164 or private dialing plan phone number. Valid entries are the digits 0 through 9 and the letters A through D. The plus symbol (+) is not valid. The following special characters can be entered:
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When the timer (T) character is included at the end of the destination pattern, the system collects dialed digits as they are entered—until the interdigit timer expires (10 seconds, by default)—or the user dials the termination of end-of-dialing key (default is #).
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(Optional) Include a dial-out prefix that the system enters automatically instead of people dialing it.
string is a value from 0 to 9, and you can use a comma (,) to indicate a pause.
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This command associates the dial peer with a specific logical interface.
slot is the router location where the voice port adapter is installed. Valid entries are from 0 to 3.
port indicates the voice interface card location. Valid entries are 0 or 1.
Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
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Enter dial-peer configuration mode and define a remote VoIP dial peer.
number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647.
voip indicates a VoIP peer using voice encapsulation on the IP network.
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Router(config-dialpeer)# codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729r8 [pre-ietf] | g729br8 } [bytes]The voice-card configuration codec command sets the codec options that are available when you execute this command. See Step 2 of the "Configuring the DSPfarm Interface" section.
If you do not set codec complexity, g729r8 with IETF bit-ordering is used.
If you set codec complexity to high, the following options are available:
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If you set codec complexity to medium, the following options are valid:
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The optional bytes parameter sets the number of voice data bytes per frame. Acceptable values are from 10 to 240 in increments of 10 (for example, 10, 20, 30, and so on). Any other value is rounded down (for example, from 236 to 230).
If you specify g729r8, then the IETF (Internet Engineering Task Force) bit-ordering is used. For interoperability with a Cisco 7200 series router running a Cisco IOS release prior to Release 12.0(5)T or12.0(4)XH, you must specify the additional key word pre-ietf after g729r8.
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(Optional) This setting is enabled by default. It activates voice activity detection (VAD). VAD allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise.
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(Optional) Dual-tone multifrequency (DTMF) describes the tone that sounds in response to a keypress on a touch-tone phone. DTMF tones are compressed at one end of a call and decompressed at the other end.
If a low-bandwidth codec, such as a G.729 or G.723, is used, the tones can sound distorted. The dtmf-relay command transports DTMF tones generated after call establishment out-of-band by using a method that transmits with greater fidelity than is possible in-band for most low-bandwidth codecs. Without DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated phone menu systems, such as voicemail and interactive voice response (IVR) systems.
A signaling method is supplied only if the remote end supports it, and the options are: Cisco proprietary (cisco-rtp), standard H.323 (h245-alphanumeric), and H.323 standard with signal duration (h245-signal).
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(Optional) Specify the transmission speed of a fax to be sent to this dial peer. disable turns off fax transmission capability, and voice specifies the highest possible fax speed supported by the voice rate.
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See Step 3 in this procedure.
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Configure the IP session target for the dial peer.
ipv4:destination-address indicates IP address of the dial peer.
dns:host-name indicates that the domain name server will resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.
There are also wildcards available for defining domain names with the keyword by using source, destination, and dialed information in the host name. For complete command syntax information, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
Verifying Voice Dial Peers
Follow the procedure below to verify dial-peer configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
Important command output is shown in bold.
Enter the privileged EXEC show dial-peer voice command. The following text is sample output from the command for a POTS dial peer:
Router# show dial-peer voice 1VoiceEncapPeer1tag = 1, dest-pat = \Q+14085551000',answer-address = \Q',group = 0, Admin state is up, Operation state is downPermission is Both,type = pots, prefix = \Q',session-target = \Q', voice-port =Connect Time = 0, Charged Units = 0Successful Calls = 0, Failed Calls = 0Accepted Calls = 0, Refused Calls = 0Last Disconnect Cause is "10"Last Disconnect Text is ""Last Setup Time = 0The following text is sample output from the show dial-peer voice command for a VoIP dial peer:
Router# show dial-peer voice 10VoiceOverIpPeer10tag = 10, dest-pat = \Q',incall-number = \Q+14087',group = 0, Admin state is up, Operation state is downPermission is Answer,type = voip, session-target = \Q',sess-proto = cisco, req-qos = bestEffort,acc-qos = bestEffort,fax-rate = voice, codec = g729r8,Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,Connect Time = 0, Charged Units = 0Successful Calls = 0, Failed Calls = 0Accepted Calls = 0, Refused Calls = 0Last Disconnect Cause is "10"Last Disconnect Text is ""Last Setup Time = 0Monitoring and Maintaining T1/E1 Digital Voice Configuration
This section presents some useful show and debugging commands for understanding, maintaining, and troubleshooting your configuration.
Table 1 debug and show Commands for Maintaining and Troubleshooting Your Configuration
Router# show dialplan number numberShows which dial-peer is matched by a called number.
Router# show call active voiceShows statistics for currently active voice calls.
Router# show call active faxShows statistics for currently active fax calls.
Router# show call history voiceShows statistics on previous voice calls.
Router# show call history faxShows statistics on previous fax calls.
Router# show voice portShows the status of voice ports. See "Verifying Voice Ports" on page 23.
Router# show controller t1 slot/portShows the status of the T1 controller. See "Verifying Card Type and Controller Settings" on page 20.
Router# show controller e1 slot/portShows the status of the E1 controller. See "Verifying Card Type and Controller Settings" on page 20.
Router# debug vpm allDebugs the T1/E1 signaling.
Router# debug vtsp allDebugs the digits received and sent.
Router# debug voip ccapi inoutDebugs the call setup process.
The balance of this section shows the output of the commands listed in .
show Commands
This section illustrates some of the privileged EXEC show commands that are useful for analyzing your system. Note that important information appears in bold, and bold text preceded by the "<<" characters explains the process.
The show dialplan number command provides information about the dial peer associated with a specified dial-plan number. Notice that the dial peer is operational and that IP Precedence has been configured to the preferred setting of 5.
Note
Cisco-router# show dialplan number 75435Macro Exp.: ##75435VoiceOverIpPeer70000information type = voice,tag = 70000, destination-pattern = `##7....',answer-address = `', preference=0,group = 70000, Admin state is up, Operation state is up,incoming called-number = `', connections/maximum = 0/unlimited,DTMF Relay = disabled,application associated:type = voip, session-target = `ipv4:171.68.253.18',technology prefix:settlement: disabledip precedence = 5, UDP checksum = disabled,session-protocol = cisco, req-qos = best-effort,acc-qos = best-effort,fax-rate = 14400, payload size = 20 bytescodec = g729r8, payload size = 20 bytes,Expect factor = 10, Icpif = 30,signaling-type = cas,VAD = disabled, Poor QOV Trap = disabled,Connect Time = 0, Charged Units = 0,Successful Calls = 3, Failed Calls = 0,Accepted Calls = 3, Refused Calls = 0,Last Disconnect Cause is "10 ",Last Disconnect Text is "normal call clearing.",Last Setup Time = 344813.Matched: ##75435 Digits: 3Target: ipv4:171.68.253.18The show call active voice command displays information about a current call:
Cisco-router# show call active voiceGENERIC:SetupTime=94523746 msIndex=448PeerAddress=##73072PeerSubAddress=PeerId=70000PeerIfIndex=37LogicalIfIndex=0ConnectTime=94524043DisconectTime=94546241CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=6251TransmitBytes=125020ReceivePackets=3300ReceiveBytes=66000VOIP:ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0]RemoteIPAddress=171.68.235.18RemoteUDPPort=16580RoundTripDelay=29 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceSessionProtocol=ciscoSessionTarget=ipv4:171.68.235.18OnTimeRvPlayout=63690GapFillWithSilence=0 msGapFillWithPrediction=180 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=70 msLoWaterPlayoutDelay=30 msReceiveDelay=40 msLostPackets=0 msEarlyPackets=1 msLatePackets=18 msVAD = disabledCoderTypeRate=g729r8CodecBytes=20cvVoIPCallHistoryIcpif=0SignalingType=casThe show call history voice command shows statistics about previous calls:
sb1pbx-voip# show call history voiceGENERIC:SetupTime=94893250 msIndex=450PeerAddress=##52258PeerSubAddress=PeerId=50000PeerIfIndex=35LogicalIfIndex=0DisconnectCause=10DisconnectText=normal call clearing.ConnectTime=94893780DisconectTime=95015500CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=32258TransmitBytes=645160ReceivePackets=20061ReceiveBytes=401220VOIP:ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C]RemoteIPAddress=171.68.235.18RemoteUDPPort=16552RoundTripDelay=23 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceSessionProtocol=ciscoSessionTarget=ipv4:171.68.235.18OnTimeRvPlayout=398000GapFillWithSilence=0 msGapFillWithPrediction=1440 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=97 msLoWaterPlayoutDelay=30 msReceiveDelay=49 msLostPackets=1 msEarlyPackets=1 msLatePackets=132 msVAD = disabledCoderTypeRate=g729r8CodecBytes=20cvVoIPCallHistoryIcpif=0SignalingType=casdebug Commands
This section illustrates some of the EXEC mode debug commands that are useful when analyzing and troubleshooting your system. Note that important information appears in bold, and bold text preceded by the "<<" characters explains the process.
The debug vpm all command displays information that allows you to troubleshoot T1/E1 signaling:
Cisco-router# debug vpm allApr 19 19:18:54 PDT: htsp_process_event: [1/0/16, 1.4 , 34] em_onhook_offhookem_offhookem_onhookhtsp_setup_ind << port goes offhookApr 19 19:18:54 PDT: htsp_process_event: [1/0/16, 1.5 , 8]Apr 19 19:19:01 PDT: htsp_process_event: [1/0/16, 1.5 , 10] htsp_alert_notifyApr 19 19:19:01 PDT: htsp_process_event: [1/0/16, 1.5 , 11]Apr 19 19:19:02 PDT: htsp_process_event: [1/0/16, 1.5 , 11]Apr 19 19:19:15 PDT: htsp_process_event: [1/0/16, 1.5 , 22] em_offhook_onhookem_stop_timers em_onhook << port goes onhookApr 19 19:19:15 PDT: htsp_process_event: [1/0/16, 1.4 , 7] em_onhook_releaseem_onhookThe debug vtsp all command displays information that allows you to troubleshoot digits received and sent on a call:
cisco-router# debug vtsp allApr 19 19:21:55 PDT: dsp_cp_tone_on: [1/0:1 (9502)] packet_len=30 channel_id=1 packet_id=72 tone_id=3 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=4000 amp_of_second=4000 direction=1 on_time_first=65535 off_time_first=0 on_time_second=65535 off_time_second=0 << providing dialtoneApr 19 19:21:59 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2,rtp_timestamp=0xF2D37240act_report_digit_beginApr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2, duration=102act_report_digit_endApr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71Apr 19 19:22:00 PDT: vtsp_timer: 34838705Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=3,rtp_timestamp=0xF2D37240act_report_digit_beginApr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=3, duration=92act_report_digit_endApr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71Apr 19 19:22:00 PDT: vtsp_timer: 34838724Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=1,rtp_timestamp=0xF2D37240 act_report_digit_beginApr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=1, duration=92act_report_digit_endApr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71Apr 19 19:22:00 PDT: vtsp_timer: 34838744Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=9,rtp_timestamp=0xF2D37240act_report_digit_beginApr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=9, duration=102act_report_digit_endApr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71Apr 19 19:22:00 PDT: vtsp_timer: 34838768Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=8,rtp_timestamp=0xF2D37218act_report_digit_beginApr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=8, duration=107act_report_digit_end*** The Caller dialed the digits 23198 ***The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call-session application and the underlying network-specific software.
During the capabilities exchange shown in the command output, both sides agree on what compression to use, and the debug voip ccapi inout output helps you determine what each side is negotiating.
You can use the output from this command to understand how calls are being handled by the router. This command shows how a call flows through the system. By using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs:
cisco-router# debug voip ccapi inoutApr 19 19:23:11 PDT: sess_appl: ev(19=CC_EV_CALL_SETUP_IND), cid(9504), disp(0) << a new call is originatingApr 19 19:23:11 PDT: ccCallSetContext (callID=0x2520, context=0x61C0806C)Apr 19 19:23:11 PDT: ccCallSetupAck (callID=0x2520)Apr 19 19:23:11 PDT: ccGenerateTone (callID=0x2520 tone=8) << dialtoneApr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520, digit=2, flags=0x1, timestamp=0xCE2796D1, expiration=0x0) << digit 2 receivedApr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=2, duration=102)Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520, digit=3, flags=0x1, timestamp=0xCE2796D1, expiration=0x0)Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=3, duration=102) << digit 3 receivedApr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520, digit=1, flags=0x1, timestamp=0xCE2796D1, expiration=0x0)Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=1, duration=92) << digit 1 receivedApr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520, digit=9, flags=0x1, timestamp=0xCE2796B9, expiration=0x0)Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=9, duration=105) << digit 9 receivedApr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520, digit=8, flags=0x1, timestamp=0xCE279691, expiration=0x0)Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=8, duration=100) << digit 8 receivedApr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)Apr 19 19:23:18 PDT: ssaSetupPeer cid(9504) peer list: tag(20000)Apr 19 19:23:18 PDT: ssaSetupPeer cid(9504), destPat(23198), matched(1), prefix(), peer(61C04464) << matched dial-peer 20000 voipApr 19 19:23:18 PDT: peer_tag=20000 << matched dial-peer voip 20000Apr 19 19:23:18 PDT: ccIFCallSetupRequest: (vdbPtr=0x61A25524, dest=, callParams << voip call setup={called=23198, calling=+9.......T, fdest=0, voice_peer_tag=20000}, mode=0x0)Apr 19 19:23:18 PDT: ccCallSetContext (callID=0x2521, context=0x61C12E18)Apr 19 19:23:18 PDT: ccCallProceeding (callID=0x2520, prog_ind=0x0)Apr 19 19:23:19 PDT: cc_api_call_alert(vdbPtr=0x61A25524, callID=0x2521, prog_ind=0x88, sig_ind=0x1)Apr 19 19:23:19 PDT: sess_appl: ev(7=CC_EV_CALL_ALERT), cid(9505), disp(0)Apr 19 19:23:19 PDT: ssa: cid(9505)st(1)oldst(0)cfid(-1)csize(0)in(0)fDest(0)-cid2(9504)st2(1)oldst2(0)Apr 19 19:23:19 PDT: ccCallAlert (callID=0x2520, prog_ind=0x88, sig_ind=0x1)Apr 19 19:23:19 PDT: ccConferenceCreate (confID=0x61A21670, callID1=0x2520, callID2=0x2521, tag=0x0)Apr 19 19:23:19 PDT: cc_api_bridge_done (confID=0x33, srcIF=0x61A25524, srcCallID=0x2521, dstCallID=0x2520, disposition=0, tag=0x0)Apr 19 19:23:19 PDT: cc_api_bridge_done (confID=0x33, srcIF=0x61A1B1B4, srcCallID=0x2520, dstCallID=0x2521, disposition=0, tag=0x0)Apr 19 19:23:19 PDT: cc_api_caps_ind (dstVdbPtr=0x61A25524, dstCallId=0x2521, sr<< negotiating capabilities with the remote VoIP gatewayApr 19 19:23:36 PDT: sess_appl: ev(8=CC_EV_CALL_CONNECTED), cid(9505), disp(0)Apr 19 19:23:36 PDT: ssa: cid(9505)st(4)oldst(1)cfid(51)csize(0)in(0)fDest(0)-cid2(9504)st2(4)oldst2(4)<< the VoIP call is connectedApr 19 19:23:54 PDT: sess_appl: ev(12=CC_EV_CALL_DISCONNECTED), cid(9505),disp(0)<< the VoIP call is disconnectedApr 19 19:23:54 PDT: ccCallDisconnect (callID=0x2520, cause=0x10 tag=0x0)<< the VoIP call is disconnected by cause_code 0x10explains the codec negotiation values that appear—in hexadecimal format— during the capabilities exchange portion of the command output.
Reference Information
The information in this section helps you interpret the output from debug and show commands.
shows Q.931 call disconnection causes. In the examples that follow, the disconnects are caused by normal call clearing.
Table 3 Q.931 Call Disconnection Causes
Table 4 Tone Types and Their Meanings
These are codec capabilities bits that can appear in command output:
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These are fax capabilities bits that can appear in command output. The numbers following "FAX_" refer to the fax speed (for example, "144" means 14,400 bps):
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These are the VAD on and off capability bits:
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Configuration Examples
This section includes the following configuration examples:
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These examples are not necessarily complete configurations. They are designed to illustrate specific tips and techniques, and only the relevant portions of the configurations are shown. Each configuration includes a brief introduction, side-by-side configurations for routers at either end, and explanations of key points.
Routed Digits — Switched VoIP Calls
Figure 6 Sample Configuration: Routed Digits
This example shows how to set up a Cisco 7200 series router to collect digits from either a PBX/PSTN or a phone and route a VoIP call based on the digits received. The commands used in the configurations are explained inline. Only relevant sections of the configuration are shown. The example assumes that the IP portion of the network is already in place.
In this configuration, the PBX seizes the T1/E1 to the router, which expects to collect digits from the PBX. Upon collecting those digits, the router tries to match a dial peer to route the call. If the router receives the correct digits, it routes the call according to the configuration of the dial peer.
Here are some key points for consideration:
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FRF.12 — Switched VoIP Calls
Figure 7 Sample Configuration: FRF.12 Switched VoIP Calls
This example shows how to configure a Cisco 7200 series router to support FRF.12 fragmentation and queuing in a voice over IP over Frame Relay network. FRF.12 is a Frame Relay Forum standard mechanism for fragmenting data packets. This fragmentation helps eliminate the delays that occur when transmitting voice and data over the same network—large data packets can delay smaller voice packets from being transmitted into the IP network. FRF.12 is also supported on the MC3810 and 7200 routers, which can be used as tandem-nodes for VoIP networks.
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This configuration fragments both the IP and IPX data traffic to 80 bytes, allowing the VoIP traffic to be only minimally delayed on the network. The FRF.12 setup also traffic-shapes the output traffic rate to match the provisioned committed information rate (CIR) from the Frame Relay carrier. This ensures that traffic is not dropped or delayed within the Frame Relay network.
Here are some key points for consideration:
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Routing Calls Through an H.323 Gatekeeper
Figure 8 Sample Configuration: Routing Calls through an H.323 Gatekeeper
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The example in this section shows how to configure a Cisco 7200 series router to route VoIP calls through an H.323 gatekeeper. This setup shows calls being routed from a gateway in Zone-Alpha, through the gatekeeper, to a gateway in Zone-Beta.
For complete documentation of H.323 gatekeeper functionality, refer to the IOS documentation on CCO at these URLs:
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Here are some key points for consideration:
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Private-Line Auto-Ringdown Configuration — Switched VoIP Calls
Figure 9 Sample Configuration: PLAR
This example shows how to set up a Cisco 7200 series router for a Private-Line auto-ringdown (PLAR). PLAR is used to allow a station or DS0 to go off-hook, and—without the user dialing digits—have a call completed to the far end. PLAR can also provide dial tone from a remote PBX for off-premises applications.
In this configuration, the phones off router Beta go off-hook and receive dialtone from the PBX connected to router Alpha. From there, users can dial any digits in to the PBX as if their stations are directly connected to it.
Here are some key points for consideration:
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Connection Trunk Configuration — Permanent VoIP Calls
Figure 10 Sample Configuration: Connection Trunk Permanent VoIP Calls
This example shows how to configure a Cisco 7200 series router for a trunk connection. A trunk connection is like a "wire" between the two routers. It is a transparent connection, so it allows features such as hookflash (also called switchhook flash) or hoot `n' holler (point-to-point) to pass. This type of trunk configuration can also be used for OPXs (Off-Premise extensions) that require rollover to a centralized voice-mail system when the user does not answer.
A trunk connection can only be used between E&M ports or with FXO-to-FXS connections.
In this configuration, a permanent and transparent path is set up between individual DS0s on each router. It passes dial-tone from the remote PBX and passes DTMF digits out of band.
The connection trunk command establishes the permanent trunk connection between the routers. The digits after the command are passed internally within the router to match a dial-peer so that the call can be set up.
Drop-and-Insert Sample Configuration
Figure 11 Sample Configuration: Drop-and-Insert
Drop-and-Insert technology is one way to integrate old PBX technologies with VoIP. With VoIP, you can take 64Kb DS0 channels from one T1/E1 and digitally cross-connect them to 64Kb DS0 channels on another T1/E1. Drop-and-Insert is sometimes called TDM cross-connect.
With Drop-and-Insert, individual 64Kb DS0 channels can be transparently passed, and uncompressed between T1/E1 ports without passing through a DSP. Using this method, the channel traffic is sent between a PBX and central office switch (PSTN) or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN, while the router compresses others for interoffice VoIP calls. In addition, Drop-and-Insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.
Here are some key points for consideration:
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Command Reference
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.0 command references.
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busyout monitor interface
To place a voice port into busyout monitor state, enter the busyout-monitor interface voice-port configuration command. To remove the busyout monitor state on the voice port, use the no form of this command.
busyout-monitor interface interface number
no busyout-monitor interface interface numberSyntax Description
Default
The voice port is not in busyout monitor state.
Command Modes
Voice-port configuration
Command History
12.0(3)T
This command was introduced for the Cisco MC3810.
12.0(5)XE
The command was modified for the Cisco 7200 series routers.
Usage Guidelines
When you place a voice port in busyout monitor state, the voice port monitors the specified interface and enters the busyout state when the interface is down. This forces rerouting of calls when an interface is down.
If you specify more than one monitored interface for a voice port, all the monitored interfaces must be down in order to trigger busyout on the voice port.
The command monitors only the up or down status of an interface—not end-to-end TCP/IP connectivity.
When an interface is operational, a busied-out voice port returns to its normal state.
This feature can monitor LAN, WAN, and virtual interfaces, as well as subinterfaces.
Examples
The following example configures the voice port to monitor two serial interfaces and an Ethernet interface. When all these interfaces are down, the voice port is busied out. When at least one interface is operating, the voice port is put back into a normal state.
voice-port 3/0:0busyout monitor interface Ethernet0/0busyout monitor interface Serial1/0busyout monitor interface Serial2/0card type
To configure the card type on the port adapter of the Cisco 7200 series router, use the card type global configuration command. Use the no form of this command to restore the default value.
card type <t1 | e1> <slot> [bay]
no card typeSyntax Description
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Examples
The following example configures T1 data transmission on port 1 on the Cisco 7200 series router:
t1 1codec (dial-peer)
To specify the voice coder rate of speech for a VoIP dial peer, enter the codec dial-peer configuration command. Use the no form of this command to restore the default value.
codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729r8 [pre-ietf] | g729br8 } [bytes]
no codec
Syntax Description
Default
The default is g729r8.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
This command applies only to VoIP dial peers.
A specific codec type can be configured on the dial peer as long as it is supported by the setting used with the codec voice-card configuration command.
The dial-peer configuration command is particularly useful when you must change to a small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link. However, g711alaw and g711ulaw provide higher-quality voice transmission than other codecs. g729r8, which provides near-toll quality with considerable bandwidth savings.
If codec values for the VoIP peers of a connection do not match, the call fails.
You can change the payload of each VoIP frame by using the byte setting. However, increasing the payload size can add processing delay for each voice packet.
Examples
The following example configures a dial peer to use the g723r53 (G.723.1 at 5,300 bps) codec type:
dial-peer voice 1 voipcodec g723r53Related Commands
codec
Based on the codec standard you are using, enter the codec DSP interface dsp farm command to specify call density and codec complexity. High-complexity codecs support lower call density than do medium-complexity codecs. The no form of the command resets the card type to the default.
codec {high | low | medium}
no codec
Syntax Description
high
Specifies high complexity: Two channels of any mix of codec.
low
Specifies low complexity: Eight channels of g711.
medium
Specifies medium complexity: Four channels of g711/g726/g729a/fax
Defaults
Medium.
Command Modes
DSP interface dsp farm
Command History
Usage Guidelines
Codec complexity refers to the amount of processing required in order to perform compression. Codec complexity affects the number of calls that can take place on the DSPfarm interfaces, referred to as call density. The greater the codec complexity, the fewer calls are handled. For example, G.711 requires less DSP processing than G.728, so that as long as the bandwidth is available, more calls can be handled simultaneously by using the G.711 standard than using G.728.
The DSPinterface dspfarm codec complexity setting affects the options available for the codec dial-peer configuration command.
To change codec complexity, you must first remove any configured CAS or DS0 groups, and then reinstate them after the change.
Examples
The following example configures the DSPfarm interface 1/0 on the Cisco 7200 series routers to support high compression:
dspint dspfarm 1/0 codec high 0-30Related Commands
command-type
Command used to search online to find documentation of related commands.
description
To include a specific description about the DSP interface, use the description configuration command. Use the no form of this command to disable this feature.
description string
no descriptionSyntax Description
Defaults
Enabled with a null string.
Command Modes
Configuration.
Command History
Usage Guidelines
Use the description command to include descriptive text about this DSP interface connection. This information is displayed when you issue a show command and does not affect the operation of the interface in any way.
Examples
The following example identifies DSPfarm interface 1/0 on the Cisco 7200 series router as being connected to the marketing department:
dspint dspfarm 1/0 description marketing_deptdspfarm
To enable the DSP interface, use the dspfarm configuration command.
dspfarm slot/port
Syntax Description
Defaults
No default behavior or values.
Command Modes
Configuration
Command History
Examples
The following example creates a DSPfarm interface with a slot number of 1 and a port number of 0.
dspint dspfarm 1/0Related Commands
ds0-group
To define T1/E1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, enter the ds0-group controller configuration command. The no form of the command removes the group and signaling setting.
ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
no ds0-group ds0-group-no
Syntax Description
Default
There is no DS0 group.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco 7200 series routers: slot/port:ds0-group-no. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Examples
The following example configures ranges of T1 controller timeslots for FXS ground-start and FXO loop-start signaling:
controller T1 1/0cablelength long 0dbds0-group 1 timeslots 4-5 type e&m-immediate-startRelated Command
codec
To change codec complexity by using this voice-card configuration command, you must first remove any configured CAS or DS0 groups; then, reinstate them after the change.
echo-cancel coverage
To adjust the maximum duration to cancel, use the echo-cancel coverage voice-port configuration command. Use the no form of this command to reset this command to the default value.
echo-cancel coverage {8 | 16 | 24 | 32}
no echo-cancel coverage
Syntax Description
Default
16 milliseconds
Command Modes
Voice-port configuration
Command History
11.3(1)T
The command was introduced.
12.0(5)XK
The command was modified to add the 8-millisecond option.
Usage Guidelines
Use the echo-cancel coverage command to adjust the coverage size of the echo canceller. This command enables cancellation of voice that is sent out the interface and received back on the same interface within the configured amount of time. If the local loop (the distance from the interface to the connected equipment producing the echo) is longer, the configured value of this command should be extended.
If you configure a longer value for this command, it takes the echo canceller longer to converge; in this case, the user might hear slight echo when the connection is initially set up. If the configured value for this command is too short, the user might hear some echo for the duration of the call because the echo canceller is not cancelling the longer delay echoes.
There is no echo or echo cancellation on the network (for example, non-POTS) side of the connection.
Note
Examples
The following example adjusts the size of the echo canceller to 8 milliseconds on the Cisco 7200 series routers:
voice-port 1/0:0echo-cancel enableecho-cancel coverage 8Related Command
reset
To reset a set of DSPs, use the reset configuration command.
reset number
Syntax Description
Defaults
No default behavior or values.
Command Modes
Configuration
Command History
Examples
The following example displays the reset command configuration for DSP 1:
reset 101:24:54:%DSPRM-5-UPDOWN: DSP 1 in slot 1, changed state to upshow interface dspfarm
To display DSP information on the two-port T1/E1 high density port adapter, use the show interface dspfarm EXEC command.
show interface dspfarm [slot/port]
The local TDM cross-connect map can be displayed by the following show command.
show int dspfarm <x/y | x/y/z> dsp tdm
Syntax Description
Defaults
No default behavior or values.
Command Modes
EXEC configuration
Command History
Examples
The following example is sample output from the show interface dspfarm command in chassis slot 3, in port adapter slot 0:
Router# show interface dspfarm 3/0DSPfarm3/0 is up, line protocol is upHardware is VXC-2T1/E1MTU 256 bytes, BW 12000 Kbit, DLY 0 usec,reliability 255/255, txload 4/255, rxload 1/255Encapsulation VOICE, loopback not setC549 DSP Firmware Version:MajorRelease.MinorRelease (BuildNumber)DSP Boot Loader:255.255 (255)DSP Application:4.0 (3)Medium Complexity Application:3.2 (5)High Complexity Application:3.2 (5)Total DSPs 30, DSP0-DSP29, Jukebox DSP id 30Down DSPs:noneTotal sig channels 120 used 24, total voice channels 120 used 00 active calls, 0 max active calls, 0 total calls30887 rx packets, 0 rx drops, 30921 tx packets, 0 tx frags0 curr_dsp_tx_queued, 29 max_dsp_tx_queuedLast input never, output never, output hang neverLast clearing of "show interface" counters neverQueueing strategy:fifoOutput queue 0/0, 0 drops; input queue 0/75, 0 drops5 minute input rate 13000 bits/sec, 94 packets/sec5 minute output rate 193000 bits/sec, 94 packets/sec30887 packets input, 616516 bytes, 0 no bufferReceived 0 broadcasts, 0 runts, 0 giants, 0 throttles0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort30921 packets output, 7868892 bytes, 0 underruns0 output errors, 0 collisions, 0 interface resets0 output buffer failures, 0 output buffers swapped out
show voice port
To display configuration information about a specific digital voice port, enter the show voice port privileged EXEC command.
show voice port slot/port:ds0-group
Syntax Description
Default
There is no default.
Command Modes
Privileged EXEC
Command History
11.3(1)T
The command was introduced.
12.0(5)XE
Additional syntax was created for digital voice to allow specification of the DS0 group.
Usage Guidelines
This command applies to VoIP on the Cisco 7200 series.
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco 7200 series routers: slot/port:ds0-group-no. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Examples
The following displays voice port configuration information for the digital voice port 0 located in slot 1, DS0 group 1:
cisco-router# show voice port 1/0:1receEive and transMit Slot is 1, Sub-unit is 0, Port is 1Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 DBMSIn Gain is Set to 0 dBmOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for USRelated Command
ds0-group
Defines T1/E1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN.
shut
To shutdown a set of DSPs, use the shut configuration command. Use the no form of this command to put DSPs back in service.
shut
no shutSyntax Description
This command has no arguments or keywords.
Defaults
No shutdown.
Command Modes
DSP configuration.
Command History
Usage Guidelines
This command applies to Voice over IP on the Cisco 7200 series routers.
Examples
The following example shuts down two sets of DSPs:
router(config-dspfarm)#shut 2Glossary
AAL—ATM Adaptation Layer. Service-dependent sublayer of the data link layer. The AAL accepts data from different applications and presents it to the ATM layer in the form of 48-byte ATM payload segments. AALs consist of two sublayers: convergence sublayer (CS) and segmentation and reassembly (SAR). AALs differ on the basis of the source-destination timing used, whether they use constant bit rate (CBR) or variable bit rate (VBR), and whether they are used for connection-oriented or connection less mode data transfer. At present, the four types of AAL recommended by the ITU-T are AAL1, AAL2, AAL3/4, and AAL5.
AAL1—ATM adaptation layer 1. One of four AALs recommended by the ITU-T. AAL1 is used for connection-oriented, delay-sensitive services requiring constant bit rates, such as uncompressed video and other isochronous traffic.
AMI—alternate mark inversion. Line-code type used on T1 and E1 circuits. In AMI, zeros are represented by 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream. Sometimes called binary coded alternate mark inversion.
ATM—Asynchronous Transfer Mode. International standard for cell relay in which multiple service types (such as voice, video, or data) are conveyed in fixed-length (53-byte) cells. Fixed-length cells allow cell processing to occur in hardware, thereby reducing transit delays. ATM is designed to take advantage of high-speed transmission media such as E3, SONET, and T3.
B8ZS—binary 8-zero substitution. Line-code type, used on T1 and E1 circuits, in which a special code is substituted whenever 8 consecutive zeros are sent over the link. This code is then interpreted at the remote end of the connection. This technique guarantees ones density independent of the data stream.
CAS—channel-associated signaling. Trunk signaling (for example, in a T1 line) in which control signals, such as those for synchronizing and bounding frames, are carried in the same channel along with voice and data signals.
CBR—constant bit rate. QoS class defined by the ATM Forum for ATM networks. CBR is used for connections that depend on precise clocking to ensure undistorted delivery.
CCS—common channel signaling. Trunk signaling (for example, using Primary Rate Interface) in which a control channel carries signaling for separate voice and data channels.
CES—circuit emulation service. Enables users to multiplex or concentrate multiple circuit emulation streams for voice and video with packet data on a single high-speed ATM link without a separate ATM access multiplexer.
CO—central office. Local telephone company office to which all local loops in a given area connect and in which circuit switching of subscriber lines occurs.
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.
DTMF—Dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).
Drop-and-Insert—(also called TDM cross connect) Allows DSO channels from one T1 or E1 facility to be digitally cross-connected to DS0 channels on another T1 or E1. Using this method, channel traffic is sent between a PBX and CO PSTN switch or other telephony device, so that some PBX channels are directed for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, Drop-and-Insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank for external analog connectivity.
DSP—digital signal processor, same as PVDM
E1—European digital carrier facility used for transmitting data through the telephone hierarchy. The transmission rate for E1 is 2.048 megabits per second (Mbps).
E&M—rEceive and transMit, or Ear and Mouth. Type of signaling originally developed for analog two-state voltage telephony using the ear and mouth leads; in digital telephony, uses two bits.
ESF—Extended Superframe. Framing type used on T1 circuits that consists of 24 frames of 192 bits each, with the 193rd bit providing timing and other functions. ESF is an enhanced version of SF format.
FXO—Foreign Exchange Office. A voice interface emulating a PBX trunk line to a switch or telephone equipment to a PBX extension interface.
FXS—Foreign Exchange Station. A voice interface for connecting telephone equipment, emulates the extension interface of a PBX or the subscriber interface for a switch.
IETF—Internet Engineering Task Force
ISDN—Integrated Services Digital Network. Communication protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other source traffic.
IVR—interactive voice response. Term used to describe systems that provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words or more commonly DTMF signaling. Examples include banks that allow you to check your balance from any telephone and automated stock quote systems.
packet—Logical grouping of information that includes a header containing control information and (usually) user data. Packets are most often used to refer to network layer units of data.
POTS—plain old telephone service
PSTN—Public Switched Telephone Network. General term referring to the variety of telephone networks and services in place worldwide.
QoS—quality of service. Measure of performance for a transmission system that reflects its transmission quality and service availability.
SF—Super Frame. Common framing type used on T1 circuits. SF consists of 12 frames of 192 bits each, with the 193rd bit providing error checking and other functions. SF is superseded by ESF, but is still widely used. Also called D4 framing.
SNMP—Simple Network Management Protocol. Network management protocol used almost exclusively in TCP/IP networks. SNMP provides a means to monitor and control network devices, and to manage configurations, statistics collection, performance, and security.
T1—Digital WAN carrier facility. T1 transmits DS 1-formatted data at 1.544 Mbps through the telephone switching network, using alternate mark inversion or B8ZS coding.
T1 trunk—Digital WAN carrier facility. See T1.
TDM—time-division multiplexing
Trunk—Physical and logical connection between two switches across which network traffic travels. A backbone is composed of a number of trunks.
UNI—User-Network Interface. ATM Forum specification that defines an interoperability standard for the interface between ATM-based products (a router or an ATM switch) located in a private network and the ATM switches located within the public carrier networks. Also used to describe similar connections in Frame Relay networks.
VAD—voice activity detection
