Table Of Contents
MIX-Enabled Multichannel T1/E1 Port Adapter
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring Card Type and T1 Controller Settings
Configuring Card Type and E1 Controller Settings
Monitoring and Maintaining MIX Connections
Drop-and-Insert of Voice or Data
MIX-Enabled Multichannel T1/E1 Port Adapter
The Multiservice Interchange (MIX) port adapter (PA-MCX) adds time-division multiplexing (TDM) connection capabilities so that you can combine types of traffic traveling through Cisco 7200 VXR series routers. On a Cisco 7200 VXR series router, MIX enables TDM connections between two ports on the same MIX-enabled port adapter.
MIX functions permit connection of TDM streams to support applications that are sensitive to time delay, such as voice and video, and provide customers the flexibility to manage this traffic through the router as traditional TDM connections or in a packet-based format. This document includes the following sections:
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Supported Standards, MIBs, and RFCs
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Monitoring and Maintaining MIX Connections
Feature Overview
The PA-MCX port adapters are multichannel port adapters that provide T1/E1 connectivity for data and voice traffic. The PA-MCX ports can be configured either as DS1/PRI ports for data traffic, or as packet voice ports that, when used in conjunction with a digital voice port adapter, allow Cisco 7200 VXR routers to become dedicated packet voice hubs or packet voice gateways that connect to both private branch exchanges (PBXs) and the Public Switched Telephone Network (PSTN). This allows packet voice and packet fax calls to be placed over the WAN and sent through the gateway into the traditional circuit-switched voice infrastructure.
The following port adapters are currently MIX-enabled:
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PA-MCX-2TE1
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PA-MCX-4TE1
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PA-MCX-8TE1
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PA-VXC-2TE1
MIX functionality is used to support the extended drop-and-insert application, and the DSP Farm feature which are described below:
Drop-and-Insert Feature
Drop-and-insert capability allows digital signal level 0 (DS0s) from several T1/E1 interfaces (such as voice, data, or video) to be aggregated on one channelized interface and routed across the network or handed off to a Public Switched Telephone Network (PSTN). Previously, separate T1/E1s were required for data and voice streams, but with MIX functionality both types of traffic can be handled in separate channels on the same T1/E1 carrier. For most operations, this means fewer T1/E1s are needed, thereby reducing operating costs.
Another benefit of drop-and-insert functionality is an increase in the flexibility and choice that administrators can exercise over different types of traffic in their networks. With the availability of both TDM and packet streams on the Cisco 7200 VXR series routers, all voice, video, and data can be routed over the same platform, thus keeping infrastructure costs down.
Time-sensitive traffic such as video and voice can be managed as TDM streams, with traditional quality of service (QoS) considerations. All the slots participating in the drop-and-insert function are synchronized across the TDM backplane to the same clock to prevent clock slips that can cause frame loss. To synchronize the data flow of these delay-sensitive data types, clocking can be obtained from internal or external sources. Gradually, voice and video traffic can be migrated to packet-based formats without having to invest in new equipment.
Drop-and-insert technology is one way to integrate old PBX technologies with packetized voice. With drop and insert, you can take 64 Kb DS0 channels from one T1/E1 and digitally cross connect them to 64 Kb DS0 channels on another T1 or E1. Drop-and-insert functionality is sometimes called TDM cross connect.
With drop-and-insert functionality, individual 64 Kb DS0 channels can be transparently passed and uncompressed between T1or E1 ports without passing through a DSP. Using this method, the channel traffic is sent between a PBX and central office switch 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 central office [CO] or PSTN) to a channel bank to provide external analog connectivity.
DSP Farm Feature
The DSP Farm feature enables a router to become an integrated voice and data solution providing port adapter support to more voice terminations. The DSP-farm capabilities of the digital voice port adapters provide a high-density voice and fax gateway solution to connect PBXs, the PSTN, and the WAN to create a multiservice network.
With MIX functionality, DSP resources are pooled into a farm so that they can be used by TDM streams regardless of the slot or port from which the streams originate. TDM streams carrying voice are routed to the appropriate DSP resources and packetized voice is sent out over the data network.
Membership in the DSP farm is optional. You can include a DSP resource in a DSP farm or reserve it to be used only on traffic from its own interfaces.
Benefits
The MIX-enabled 2/4/8 Port Multichannel T1/E1 port adapter provides the following benefits:
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Large scale voice and data termination—A maximum of 48 T1/E1data interfaces can be supported on a Cisco 7206 VXR router.
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Drop-and-Insert functionality—DS0 drop-and-insert functionality between any interface within the port adapter provides efficient traffic aggregation and reduction in WAN links.
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DSP Farm—PA-VXC-2TE1+ port adapters in Cisco 7200 VXR series routers can share DSP resources providing low-cost, high-density voice termination. Twenty T1 or sixteen E1 voice interfaces can be terminated on a single Cisco 7206 VXR router.
Related Features and Technologies
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PA-MC-T1—in data mode
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PA-MC-E1—in data mode
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PA-VXC-TE1+
Related Documents
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MIX Enabled Multichannel T1/E1 Port Adapter Installation and Configuration
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Configuring T1/E1 High Capacity Digital Voice Port Adapters
Supported Platforms
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Cisco 7200 VXR series routers
Supported Standards, MIBs, and RFCs
Standards
The following standards are supported by the PA-MCX when used together with a PA-VXC-2TE1 for voice applications only:
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FRF.11 Voice over Frame Relay
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H.323 Version 2 Voice over IP including RAS, Fast Connect, codec negotiation
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ITU standard compression algorithms (G.703, G.704, G.729, G.723.1, G.729a/b, G.711, G.728, G.726)
MIBs
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CISCO-VOICE-DIAL-CONTROL-MIB
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB web site on Cisco Connection Online (CCO) at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
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RFC1406, Definitions of Managed Objects for the DS1 and E1 Interface Types
Prerequisites
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Multiservice Interchange (MIX) card installed on a Cisco 7200 VXR series router
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Cisco IOS Release 12.1(5)T installed
Configuration Tasks
See the following sections for configuration tasks for the MIX port adapter feature. Each task in the list is identified as either optional or required.
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Configuring Card Type and T1 Controller Settings (Required)
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Configuring Card Type and E1 Controller Settings (Required)
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Configuring the Service Type (Required)
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Configuring a Voice Port (Required only for voice applications)
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Configuring Voice Dial Peers (Required for voice applications only)
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Configuring Network Clock Use (Required only for voice applications)
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.
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.
Configuring the Service Type
Because the PA-MXC can be used for either voice or data connections, you must configure the T1/E1 controller for either data or voice using the service type controller configuration command:
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Configuring a Voice Port
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 12.1.
Configuring 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 12.1.
Command PurposeStep 1
Router# configure terminalEnter global configuration mode.
Step 2
Router(config)# dial-peer voice number potsEnter dial-peer configuration mode and define a local dial peer that will connect to the plain old telephone service (POTS) network.
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number—one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647.
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pots—a peer using basic telephone service.
Step 3
Router(config-dialpeer)# destination-pattern string [T]Configure the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number.
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string—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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The star character (*) that appears on standard touch-tone dial pads can be in any dial string but not as a leading character (for example, *650).
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The period (.) acts as a wildcard character.
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The comma (,) can be used only in prefixes and inserts a one-second pause.
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 #).
Note
The timer character must be a capital T.
Step 4
Router(config-dialpeer)# prefix string(Optional) Include a dial-out prefix that the system enters automatically instead of people dialing it.
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string—a value from 0 to 9, and you can use a comma (,) to indicate a pause.
Note
There are other digit manipulation commands available to handle such situations as prefixes for special services, ignoring some digits, and dialing into remote PBXs as though they are local.
Step 5
Router(config-dialpeer)# port slot/port:ds0-group-noThis command associates the dial peer with a specific logical interface.
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slot—the router location where the voice port adapter is installed. Valid entries are from 0 to 3.
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port—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.
Step 6
Router(config)# dial-peer voice number voipEnter dial-peer configuration mode and define a remote VoIP dial peer.
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number—one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647.
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voip—a VoIP peer using voice encapsulation on the IP network.
Step 7
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. 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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g711alaw—G.711 A Law 64,000 bps
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g711ulaw—G.711 u Law 64,000 bps
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g723ar53—G.723.1 Annex A 5,300 bps
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g723ar63—G.723.1 Annex A 6,300 bps
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g723r53—G.723.1 5,300 bps
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g723r63—G.723.1 6,300 bps
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g726r16—G.726 16,000 bps
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g726r24—G.726 24,000 bps
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g726r32—G.726 32,000 bps
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g728—G.728 16,000 bps
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g729r8—G.729 8,000 bps (default)
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g729br8—G.729 Annex B 8,000 bps
If you set codec complexity to medium, the following options are valid:
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g711alaw—G.711 A Law 64,000 bps
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g711ulaw—G.711 u Law 64,000 bps
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g726r16—G.726 16,000 bps
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g726r24—G.726 24,000 bps
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g726r32—G.726 32,000 bps
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g729r8—G.729 Annex A 8,000 bps
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g729br8—G.729 Annex B with Annex A 8,000 bps
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.
Step 8
Router(config-dialpeer)# vad(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.
Step 9
Router(config-dialpeer)# dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric](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 voice mail 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).
Step 10
Router(config-dialpeer)# fax-rate {2400 | 4800 | 7200 | 9600 | 12000 | 14400 | disable | voice}(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.
Step 11
Router(config-dialpeer)# destination-pattern string [T]See Step 3 in this procedure.
Step 12
Router(config-dialpeer)# session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name}Configure the IP session target for the dial peer.
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ipv4:destination-address—IP address of the dial peer.
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dns:host-name—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 12.1.
Configuring Network Clock Use
Time delay-sensitive traffic such as voice and video requires cross-connected port adapters to synchronize their data flow. When you configure network clocking, you can select which modules participate in clocking that is synchronized across the TDM backplane and what the source of that network clock will be. You can specify several levels of priority, so if your highest priority is unavailable, your second choice takes over automatically. Potential clock sources include internal generation from a port adapter phase-locked loop (PLL) or external retrieval from PSTN or PBX clocks over trunk lines.
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Troubleshooting Tips
The table below lists debug and show commands to help analyze problems with your configuration.
The following example shows output for the show diag command for slot 2:
Router# show diag 2Slot 2:PA-MCX-8TE1 Port adapter, 8 portsPort adapter is analyzedPort adapter insertion time 6d00h agoEEPROM contents at hardware discovery:Hardware Revision : 1.0PCB Serial Number : MIC04291ZS4Part Number : 73-4118-03Board Revision : 01RMA Test History : 00RMA Number : 0-0-0-0RMA History : 00Deviation Number : 0-0Product Number : PA-MCX-8TE1Top Assy. Part Number : 800-05358-03EEPROM format version 4EEPROM contents (hex):0x00: 04 FF 40 01 7A 41 01 00 C1 8B 4D 49 43 30 34 320x10: 39 31 5A 53 34 82 49 10 16 03 42 30 31 03 00 810x20: 00 00 00 00 04 00 80 00 00 00 00 CB 94 50 41 2D0x30: 4D 43 58 2D 38 54 45 31 20 20 20 20 20 20 20 200x40: 20 C0 46 03 20 00 14 EE 03 FF FF FF FF FF FF FF0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF0x60: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 000x70: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00Monitoring and Maintaining MIX Connections
To monitor MIX settings and connections, use the following commands.
Configuration Examples
This section provides a configuration example of drop-and-insert of voice or data.
Drop-and-Insert of Voice or Data
This example shows drop-and-insert of time slots 13-24 of T1 1/0 to time slots 1-12 of T1 1/1. The TDM groups are defined by the tdm-group command. The drop-and-insert connection is made by the connect tdm command. The time slots can carry either voice or data. When they carry voice in CAS applications, the service-type cas-voice controller configuration command must be issued first.
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The tdm-group 2 timeslots 13-24 command defines drop-and-insert capability by setting up the time slots from each T1/E1 that are used in the digital cross-connect.
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The connect tdm1 command activates the drop-and-insert digital cross connect between the T1s. The tdm1 portion of the command is a name for the cross connect, and the name can be any word, number, or series of letters.
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You can verify drop-and-insert connections by using the show connect command.
hostname RTR-A!card type t1 1!controller T1 1/0service-type cas-voiceclock source lineframing esflinecoding b8zsds0-group 1 timeslots 1-12 type e&m-winktdm-group 2 timeslots 13-24!controller T1 1/1service-type cas-voiceclock source lineframing esflinecoding b8zstdm-group 3 timeslots 1-12!!connect tdm1 T1 1/0 2 T1 1/1 3Command Reference
This section documents new commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command reference publications.
reserve
To specify how many DSP resources on a DSP farm-type interface are locally reserved, use the reserve DSP interface configuration command. To disable reservation, us the no form of this command.
reserve 0-30
no reserve
Syntax Description
Defaults
No DSP farm-type interfaces are reserved.
Command Modes
DSP interface configuration
Command History
Usage Guidelines
Using the reserve command to reserve DSP resources on a DSP farm-type interface prevents these resources from being used by external ports. It is not generally necessary to do this.
Examples
In the following example, four DSP resources are locally reserved on a DSP farm-type interface:
Router (dsp-interface)# reserve 4service-type
To specify whether the port is to be configured as a data-only port or for a TDM cross-connect to a digital voice port adapter, use the service-type configuration controller command. To disable configuration, use the no form of this command.
service-type {data | cas-voice | ccs-voice}
no service-type {data | cas-voice | ccs-voice}
Syntax Description
data
Data-only port.
cas-voice
Voice port with CAS signalling.
ccs-voice
Voice port with CCS signalling.
Defaults
ccs-voice
Command Modes
Configuration controller
Command History
Usage Guidelines
Use the service type data command to retain the maximum compatibility with traditional multichannel port adapters such as the PA-MC-T1 and PA-MC-E1. In this mode, no voice traffic is supported. You can configure the clock source individually for each port.
Use the service type ccs-voice command to connect to telephone equipment that supports common channel signalling such as ISDN PRI or QSIG. All ports configured in this mode must use a common clock source to avoid framing slips. Use the frame-clock select global configuration command to configure a port as the primary clock source. Use the clock source internal controller configuration command to configure the other ports to derive an internal clock source.
Use the service type cas-voice command to connect to connect to telephone equipment that uses CAS signalling such as T1 CAS or E1 R2. When a port is set to CAS voice mode, 64Kbs data channel-groups can also be configured on that same T1 port.
CAS-Voice mode preserves ABCD signalling across channels for drop and insert applications. All ports configured in this mode must use a common clock source to avoid framing slips. Use the frame-clock select global configuration command to configure a port as the primary clock source. Use the clock source internal controller configuration command to configure the other ports to derive an internal clock source.
Examples
The following example configures the second port in slot 1 as a voice port with CAS signalling:
Router(config)# controller t1 1/1Router(config-controller)# service-type cas-voiceRelated Commands
Glossary
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 connectionless 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 independently 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.
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 signals.
Drop-and-Insert—(also called TDM cross connect) Allows DS0 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.
DTMF—Dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).
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 (AMI) 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

