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This document describes the Digital J1 Voice Interface Card feature in Cisco IOS Release 12.2(8)T. It includes the following sections:
The J1 interface card provides the proper interface for directly connecting Cisco multiservice access routers to Private Branch Exchanges (PBXs) throughout Japan that use a J1 interface (2.048 Mbps TDM interface). This interface card supports 30 voice channels per port.
It provides the software and hardware features required to connect to over 80percent of the PBXs within Japan that use digital interfaces. This new J1 voice interface card (VIC) provides a TTC JJ-20.11 compliant interface between high-density voice network modules (NM-HDV) and a Japanese PBX.
The digital J1 card provides a single-port line interface in a VIC form factor. It is specifically designed to conform to the TTC JJ-20.10-12 standards that define the interface between a PBX and time-division multiplexer (TDM).
Figure 1 shows the earlier solution offered to customers in Japan. A J1/T1 adapter box installed between the PBX and router provides the translation between J1 using coded mark inversion (CMI) line coding at a bit rate of 2.048 Mbps and a T1 line using either alternate mark inversion (AMI) or B8ZS line coding at a bit rate of 1.544 Mbps. Note that with this solution, only 24 channels are supported, instead of the full 30 channels of the J1 interface.
Figure 2 shows the solution using the J1 interface card. The interface is now between J1 and the VIC's time division multiplex access (TDMA) bus. Note that now all 30 channels of the J1 interface are supported.
Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions at http://www.cisco.com/register .
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml
See the following sections for configuration tasks for this feature. Each task in the list is identified as either required or optional:
Use the following procedure to configure the J1 controller.
| Command | Purpose | |
|---|---|---|
| Step 1 | ||
| Step 2 | Selects the J1 controller to configure. slot/port—Backplane slot number and port number on the controller. |
Configure the DS0 groups on the J1 controller for voice applications. The J1 controller supports the E&M wink start and E&M immediate channel associated signaling (CAS) protocols for the voice ports.
The following parameters have default values for the J1 interface:
| Command | Purpose | |
|---|---|---|
| Step 1 | ||
| Step 2 | Selects the J1 controller to configure and enters controller configuration mode. This example configures a J1 controller in slot 1 and port 0. |
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| Step 3 |
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Command defines the j1 1/0 for use by compressed voice calls and the signaling method the router uses to connect to the PBX. Note This step shows the basic syntax and signaling types available with the ds0-group command. For the complete syntax, see the Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2. The keywords and arguments are as follows:
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| Step 4 | Exits to global configuration mode. Return to Step 2 if your router has more than one J1 controller that you need to configure. |
Use the following procedure to configure the clock source for a J1 controller.
Use the following procedure to configure the loopback for testing a J1 controller.
| Command | Purpose | |
|---|---|---|
| Step 1 | ||
| Step 2 | Selects the J1 controller to configure. slot/port—Backplane slot number and port number on the controller. |
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| Step 3 |
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This command defines the j1 0 for use by compressed voice calls and the signaling method the router uses to connect to the PBX. Note This step shows the basic syntax and signaling types available with the ds0-group command. For the complete syntax, see the Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2. The keywords and arguments are as follows:
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| Step 4 | ||
| Step 5 | ||
| Step 6 | Enters dial-peer configuration mode and define a local dial peer that connects to the plain old telephone service (POTS) network. The value of number is one or more digits identifying the dial peer. Valid entries are from 1 through 2147483647. The pots argument indicates a peer using a basic telephone service. |
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| Step 7 | Configures the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number. The value of 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. You can enter the following special characters: 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. |
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| Step 8 | Associates the dial peer with a specific logical interface. The value of slot is the router location where the voice module is installed. Valid entries are from 0 through 3. The value of 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. |
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| Step 9 | Exit dial-peer configuration mode to complete the POTS dial-peer configuration. |
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| Step 10 | Enters dial-peer configuration mode and defines a remote VoIP dial peer. The value of number is one or more digits identifying the dial peer. Valid entries are from 1 through 2147483647. The voip argument indicates a VoIP peer using voice encapsulation on the IP network. |
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| Step 11 | Set codec option to clear-channel to use the clear channel codec. |
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| Step 12 | (optional) This setting is enabled by default. It activates voice activity detection (VAD) which allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise. |
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| Step 13 | Configures the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number. The value of 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. You can enter the following special characters: 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. |
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| Step 14 | Configure the IP session target for the dial peer. The ipv4:destination-address parameter indicates IP address of the dial peer. The dns:host-name parameter 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 Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2 |
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| Step 15 | Exit dial peer configuration mode for the VoIP dial-peer configuration. |
Use the following procedure to configure transparent common channel signaling (T-CCS).
Three digital loopback modes are possible for diagnostics and fault isolation.
The J1 Framer has three loopback modes that are initiated through software control; line loopback, local loopback, and isolation loopback. Line loopback loops the received signal (R-D) from the PBX to the transmit going back to the PBX. Local loopback loops the transmitted signal (T-D) from the host to the receive going back to the host. Isolation loopback routes PBX and TDM generated traffic back to their respective sources. (Tx=transmit interface; Rx=receive interface;
Tip / Ring leads carry audio between the signaling unit and the trunking circuit).
| Command | Purpose |
|---|---|
Line loopback loops the receiver inputs to the transmitter outputs. The receive path is not affected by the activation of this loopback. |
| Command | Purpose |
|---|---|
Local loopback loops the transmit line encoder outputs to the receive line encoder inputs. The transmit path is not affected by the activation of this loopback. |
To monitor and maintain the J1 controller use the following privileged EXEC command.
| Command | Purpose |
|---|---|
The following displays the screen output using the show running-config command. Then it is broken down into specific examples:
The following example shows the Cisco IOS interface card in slot 4, port 0 of a Cisco 3660 configured as a J1 controller:
The following example shows the DS0 groups on the J1 controller.
The following example shows the J1 controller clock source is configured to line, where the controller recovers external clock from the line and provides the recovered clock to the internal (system) clock generator.
The following example shows the loopback method for testing the J1 controller is set at the line level.
The following example shows the codec option set to clear-channel.
This section documents new and modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications.
To display statistics about the J1 link use the show controllers j1 command in privileged EXEC mode.
Syntax Description
Defaults
No default behavior or values.
Command Modes
Command History
| Release | Modification |
|---|---|
The command was introduced on the J1 controller for the Cisco 2600 and Cisco 3600 series. |
Examples
The following example is sample output from the show controllers j1 command on the Cisco 3660:
Table 1 describes the fields shown in the display.
To configure the clock source for a J1 controller, use the clock source command in controller configuration mode. To restore the clock source to its default setting, use the no form of this command.
Syntax Description
Defaults
Clock source is line for the J1 controller.
Command Modes
Command History
| Release | Modification |
|---|---|
The command was introduced as a J1 controller configuration for the Cisco 2600 and Cisco 3600 series. |
Usage Guidelines
If multiple network modules are present in the router, then each J1 controller must be given a separate priority by configuration of the network-clock-select command. The controller having the highest priority will drive the internal clock.
Examples
The following example configures the clock source for line:
Router(config-controller)# clock source line
Related Commands
To configure a J1 controller and enter controller configuration mode, use the controller command in global configuration mode.
controller {t1 | e1 | j1} slot/port
Syntax Description
Defaults
No J1 controller is configured.
Command Modes
Command History
Examples
The following example configures the Cisco IOS interface card in slot 3, port 0 of a Cisco 3660 as a J1 controller:
Related Commands
| Command | Description |
|---|---|
To configure channelized J1 time slots enter the ds0-group command in controller configuration mode. The no form of the command removes the DS-0-group.
Syntax Description
Defaults
Command Modes
Command History
Usage Guidelines
The ds0-group command replaces the existing cas-group command. Making the command generic allows flexibility and scalability. It is not restricted to CAS signaling or channel bundling.
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco 2600 and Cisco 3600 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 is sample output from the show controllers j1 command on the Cisco 3660 series:
Related Commands
To set the loopback method for testing the J1 interface, enter the loopback command in controller configuration mode. Use the no form of this command to turn off loopback. The command should only be used for testing purposes.
Syntax Description
Defaults
Command Modes
Command History
Examples
The following example establishes a loopback of the incoming J1 signal on controller j1 3/0:
To reload the firmware and field programmable gate array (FPGA) without reloading the Cisco IOS image, use the microcode reload controller command in privileged EXEC mode.
Syntax Description
Defaults
No microcode reload activity is initiated.
Command Modes
Command History
Usage Guidelines
Configurations such as loopbacks in the running configuration are restored after this command is entered. If the controller is in a looped state before this command is issued, the looped condition is dropped. You have to re-initiate the loopbacks from the remote end by doing no loop from the controller configuration.
The following example shows the microcode reload activity being initiated:
AIS—alarm indication s22ignal. An all-ones signal transmitted in lieu of the normal signal to maintain transmission continuity and to indicate to the receiving terminal that there is a transmission fault that is located either at, or upstream from, the transmitting terminal.
AMI—alternate mark inversion. Line-code type used on T1 and E1 circuits.
CAS—channel associated signaling. The transmission of signaling information within the voice channel. CAS signaling often is referred to as robbed-bit signaling because user bandwidth is being robbed by the network for other purposes.
CCS—common channel signaling. Signaling system used in telephone networks that separates signaling information from user data. A specified channel is exclusively designated to carry signaling information for all other channels in the system.
CMI—coded mark inversion. ITU-T line coding technique specified for STS-3c transmissions.
codec—In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/decompress speech or audio signals.
E&M—recEive and transMit (or ear and mouth). Trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.
FPGA—field programmable gate array.
J1 framer—A functional block within the VIC FPGA which works in tandem with the LIUs to perform the J1 framing, monitoring and loopback functions.
MGCP—Media Gateway Control Protocol. A merging of the IPDC and SGCP protocols.
OOF—Out Of Frame. A designation for a condition defined as either the network or the DTE equipment sensing an error in framing bits.
NM-HDV—High-Density Voice network modules.
SIP—session initiation protocol. Protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.
TDM—time division multiplex. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
TDMA—time division multiplex access. Type of multiplexing where two or more channels of information are transmitted over the same link by allocating a different time interval ("slot" or "slice") for the transmission of each channel, that is, the channels take turns to use the link. Some kind of periodic synchronizing signal or distinguishing identifier usually is required so that the receiver can tell which channel is which.
VIC—voice interface card. Connects the system to either the PSTN or to a PBX.
VoATM—Voice over ATM. Voice over ATM enables a router to carry voice traffic (for example, telephone calls and faxes) over an ATM network. When sending voice traffic over ATM, the voice traffic is encapsulated using a special AAL5 encapsulation for multiplexed voice.
VoFR—Voice over Frame Relay. Voice over Frame Relay enables a router to carry voice traffic (for example, telephone calls and faxes) over a Frame Relay network. When sending voice traffic over Frame Relay, the voice traffic is segmented and encapsulated for transit across the Frame Relay network using FRF.12 encapsulation.
VoIP—Voice over IP. The ability to carry normal telephony-style voice over an IP-based internet with POTS-like functionality, reliability, and voice quality.
Posted: Wed Jan 15 10:30:10 PST 2003
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