Using the Analog Voice Module

Overview of Analog Voice Module

The analog voice module provides up to four analog telephone channels, letting you connect your telephone equipment directly to the 3000 Series. The analog voice module performs the following functions to service the telephone connection:

Voice compression and fax processing
Call state processing of the signalling information
Tone generation, including call progress, dial tone, busy signal, Dual-Tone Multi-Frequency (DTMF) ringback, and others.
Ring signal/voltage generation
Transport of voice band, signalling, and control information.
Telco power conversion, regulation, and isolation
FXO, foreign exchange office; FXS, foreign exchange station; or E&M, ear and mouth.

When the analog voice module receives analog voice and signalling information, it converts the signal to digital, compresses the signal, and then sends the signal across the network. To compress the signal, the analog voice module uses the voice call algorithm configured on the channel.

The following sections describe how to create and configure voice channels on the analog voice module.

Switched Frame Transfer Mode

SFTM adds a layer of intelligent routing, addressing, and network management to support voice/fax and data routing over IP networks. It automatically maps dial and address plans, while supporting address translations for individual lines or groups of lines.

SFTM gives you the benefits of SVC (switched virtual circuit) service using PVC (permanent virtual circuit)-based frame relay networks. Whether the traffic is voice/fax, frame relay, X.25, SNA/SDLC, asynchronous, or nearly any LAN protocol, SFTM automatically determines the best available network path for all traffic. SFTM continually monitors line outages and restorations, adjusting traffic routing accordingly with no loss of network manageability or control.

Before You Begin

To set up your analog voice module, you use the NetrixView 2000 software. This section gives you basic information on using the software and shows how to set up NetrixView to communicate with the 3000 Series. It also describes the default database that comes with the 3000 Series.

Specifically, it covers the following topics.

Assigning an IP Address to the Analog Voice Module

To communicate with the analog voice module, you need to assign an IP address to the module. To do so, use the OpenROUTE command line interface (CLI).

1. Connect to the CLI and then display the Voice Config> prompt.

*config

Config>voice 

VOICE Config>

2. Use the set address command to set the IP address.

VOICE Config> set address = 192.168.20.10

3. Restart the router to cause the new address to take affect.

VOICE Config> exit
Config>
*restart
Are you sure you want to restart the gateway? (Yes or [No]): Yes
OpenROUTE(tm) Software
OpenROUTE is a registered trademark of Netrix Corp.
Copyright Notices:
Copyright 1985-2000 by Netrix Corp., All rights reserved
Copyright 1984-1987, 1989 by J. Noel Chiappa

MOS Operator Control

*

For the remainder of this document, you will use the NetrixView 2000 software.

Using the NetrixView 2000 Software

Here is some basic information on using the NetrixView 2000 software and accessing the online help. For a detailed description on using the software, see the Network Management System User Guide

The bottom of each NetrixView 2000 screen lists the three actions that mouse buttons can perform. These actions change depending on what screen is active and where the cursor is located.

The left mouse button performs the action on the left, the middle mouse button performs the action in the center, and the right mouse button performs the action on the right. If your mouse has two buttons, click both buttons at the same time to simulate the middle button.

In this document, the actions that mouse buttons can take appear in bold. Here are some examples.

This instruction . . . means to . . .

Get Info on the NX Analog Board icon.
Move your mouse over the icon and click the right mouse button.

Select the NX Analog Board Component. Move your mouse over the component and click the left mouse button.

View into the facility.
Move your mouse over the facility composite object and click the middle mouse button.

This instruction . . .	means to . . .
Get Info on the NX Analog Board icon.	Move your mouse over the icon and click the right mouse button.
Select the NX Analog Board Component.	Move your mouse over the component and click the left mouse button.
View into the facility.	Move your mouse over the facility composite object and click the middle mouse button.

To access online help, place the cursor on the field, button, or other screen item and press the Help mouse button. The Help mouse button is usually the center button.

To close a window, or popdown to the previous level, click the right mouse button.

To edit text or numbers in a field, you must leave the cursor on the entry field. When you move the cursor from the field, the text or numbers in the field are accepted as entered, as if you pressed the Enter key.

Logging In to the NetrixView 2000

After you have installed NetrixView 2000 and set up an initial virtual network, as described in Installing the NetrixView 2000 Software, you can launch and log in to the NetrixView 2000.

The default login entries are

Network: netrix

Name: tech

Password: (Do not type a password, just press Enter.)

When you log in to NetrixView for the first time, the view is blank. The next step is to initialize the PC database.

Initializing Your NetrixView PC Database

The first time you use your NetrixView software, you need to initialize your PC database. To do so, follow these steps.

: 1. Select the Create button located at the top of the screen to put NetrixView in create mode.
: 2. On the control column, click the Goto button.
: 3. In the Node No. field, type 1 and Select View.

The following view appears, which shows various components on your PC.

4. Click each component with your left mouse button.

5. Select the Monitor button located at the top of the screen to put NetrixView in monitor mode.

6. Select each component with your left mouse button and Select the Configure button on the screen that appears.

Each component on your PC should turn green, which means that component is responsive.

Creating and Configuring an SFTM Trunk

You need to set up an SFTM trunk, which lets the NetrixView 2000 software communicate with the analog voice module. You cannot configure your module until you add this trunk on the network management system screen.

For detailed information on setting up trunks, refer to Setting Up Trunks, Chapter 6 of Configuring and Monitoring a Network Exchange 2200.

Note: Make sure the NetrixView 2000 is in Create mode.

1. Select the Composite Menu button located on the control column.

The Composite Configuration Management popup appears.

2. In the Composite Objects selection list, Select the IP SFTM Trunk composite. (You may need to scroll down through the list of objects to see it.) In the Composite Name field, type a name for the trunk. Select the Create button, and Place the icon.

The IP SFTM trunk icon is created.

Set up the Remote IP Address and Remote Node Number

1. In node view, get Info on the IP SFTM trunk icon.

The Top Level IP SFTM Trunk Info popup appears.

2. Select the UDP/IP PFD component.

The IP PFD Configuration popup appears.

3. In the Remote IP Address field, enter the IP address you assigned to the analog voice module.

4. In the Remote Node Number field, enter 3001, which is the default node number for the analog voice module.

Note: In analog voice software releases prior to r01.01.00, the default node number is 4090.

5. To update the database and implement the configuration changes, Popdown to the Top Level IP SFTM Trunk Info popup and Select the Update & Config button.

Set Up the Packet Stream Configuration for the Trunk

: 1. In node view, get Info on the IP SFTM trunk icon.
The Top Level IP SFTM Trunk Info popup appears.

: 2. Select the Packet Stream component.
The Packet Stream Configuration popup appears.
: 3. Set the CIR (bps) to 8000.
: 4. Set the Class of Service to Privileged.
: 5. To update the database and implement the configuration changes, Popdown to the Top Level IP SFTM Trunk Info popup and Select the Update & Config button.

Using the Default Analog Database

Note: The analog voice module comes with a default database that has the analog voice board and up to four channels, depending on your model. To see this default database, go to Node 3001, which is the default node number for the analog voice module. (In analog voice software releases prior to r01.01.00, the default node number is 4090.)

: 1. On the control column, click the Goto button.
: 2. In the Node No. field, type 3001 and Select View. (If you are using a software release prior to r01.01.00, type 4090 as the node number.)

Setting Up the Analog Voice Channels

The following sections describe the tasks you need to perform to set up your analog voice channels. The tasks are

You need to perform these tasks for each voice channel. You can set up templates, so that you do not have to re-enter all parameters for each channel. For information on templates, see Section 5.1.5 Working with Templates in chapter 5 of Configuring and Monitoring a Network Exchange 2200.

Configuring Standard Voice Channels

Follow the instructions in this section for each analog voice channel.

1. Get Info on the channel.

The Top Level XV Voice Channel Info popup appears.

2. To name the channel icon, Select the General Information button, type a name in the Name entry field, then Select the Update General Information button.

The name appears beneath the icon in the view.

3. To set addresses and address translation, Select the Network Interface component.

To assign an address,

a. on the Network Interface Configuration popup, Select the Addresses button.

b. On the Addressing popup that appears, which is the same as for all subscribers, enter the address for this voice channel. This is generally the dial number for this port.

Designate the address as the Main address; typically, you specify that the address can handle calls to and from the address. To do so, click on the Address Usage field with your left mouse button until Route TO and FROM Address displays.

c. Click Add and Popdown to the Network Interface Configuration popup.

To set up address translation,

a. on the Network Interface Configuration popup, Select the Address Translation button.

b. On the Address Translation popup that appears, which is the same as for all subscribers, enter one set of the following address translation rules if they are not already present. To do so, type each string in the Rule Definition field and click Add Rule.

· If this voice channel connects to a telephone, or if this channel does not need to pass the dialed digits through to the attached equipment for any other reason, use the following rules:

O:{+}=
I:{+}={1}

The outbound rule strips off the dialed network address before forwarding the call user data. The inbound rule forwards all received data, for use in directing the call.

· If this voice channel connects to a PBX or to other equipment that needs to receive the dialed digits, use the following rules:

O:{+}={1}
I:{+}={1}

The outbound and inbound rules forward all the dialed or received data to the attached equipment, for use in directing the call.

Note: For complete guidelines on creating and using address translation rules, including modifying address translation rules to fit your site, see the Network Management System User Guide.

4. To update the database and implement the configuration changes, Popdown to the Top Level XV Voice Channel Info popup and Select the Update & Config button.

Configuring the Voice Call Handler Component

The Voice Call Handler component specifies how this channel initiates and accepts a voice/fax call. The primary function of this component is to establish the destination address of an outgoing call.

You can configure the channel to be one of the following:

Dialed address channel lets you dial up a connection at any time by entering an address. This is the recommended configuration for most voice/fax channels. To set up this channel type, proceed to Configuring a Dialed Address Channel.
Automatic ring down channel connects to a specified address when the receiver goes off hook. This is the recommended configuration for STDM data channels, to connect to a remote STDM port when a scheduled Connect Request occurs. To set up this channel type, proceed to Configuring an Automatic Ring Down Channel.
Hoot-n-holler channel keeps up a constant connection to a specified address, or raises a connection to a specified address at preset times of day. To set up this channel type, proceed to Configuring a Hoot-n-Holler Channel.

Configuring a Dialed Address Channel

1. In node view, get Info on the appropriate channel.

2. Select the Voice Call Handler component.

The Voice Call Handler Configuration popup appears.

3. In the Voice Channel Type selection list, Select the Dialed Address option.

The parameters required for this channel type are activated on the popup.

4. Type the following dial plan rule in the Dial Rule Definition entry field (adjust the number of $ placeholders to reflect your dialing scheme). Enter the maximum numbers of digits that telephone numbers in your network may have.

$$$$$$$$$$$=DONE

This example tells the channel to collect 11 digits dialed and use them as the address to make the call. If you dial less than this number of digits, the channel attempts the call after the amount of time specified on the Interdigit Timeout slide lever.

Note: For security reasons, you may prefer to make more specific dial rules. See the online help on the Dial Rule Definition field for assistance and a discussion of dial rule syntax.

5. Select the Add Rule button.

The rule appears in the Dial Plan Rules selection list. If your rule syntax was incorrect or if you use lower-case letters, an error message appears.

When a user finishes dialing a call, the channel compares all dial rules to the entered digit(s), starting with the top rule on the list and working down to the last rule. The channel uses the first rule that matches the address string for that address.

6. To add other dial rules, enter the rule in the Dial Rule Definition field and then Select the Add Rule button.

7. To test the configured rules, perform the following steps:

a. Select the Test Rules button.

The Dial Plan Rule Test popup appears.

b. Select the numbers of the address on the popup's keypad. After you select each digit, the software tests all dial rules in the list with the digits accumulated thus far. It tests the top rule first, then the second rule, and so on to the bottom rule. As soon as the address matches a configured rule, that rule is highlighted. Select the Clear Test button to clear the Digits Dialed entry field.

You can also enter the digits of a possible address in the Digits Dialed entry field, and the address is tested when you press Return.

c. Popdown the Dial Plan Rule Test popup.

8. To increase or decrease the amount of time the Voice Call Handler waits between dialed digits, adjust the Interdigit Timeout slide lever. If this time period elapses before a caller enters another digit or digits match a dial rule, the call attempt is canceled and the caller hears a rapid busy tone.

9. To update the database and implement the configuration changes, Popdown the Voice Call Handler Configuration popup and Select the Update & Config button.

You are now ready to proceed to Configuring the Channel Driver Component.

This Time of Day statistic has its first event, which gives this statistic a value of 1 at the designated time.

5. Set the time you want the call taken down as follows:

In Statistic Value, select 0.
· In Hour and Minute, designate the exact time you want the call to be taken down.
· In Weekdays, Day and Month, keep the same values you used in the previous step.
· Select the Add Event button.
This Time of Day statistic has its second event, which gives this statistic a value of 0 at the designated time.

6. To update the database and implement the configuration changes, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

Defining a Hoot-n-holler Action

The following steps show how to define a hoot-n-holler action that launches a call to a specific address, and an action that ends the call.

1. On the Top Level Voice/Fax Channel Info popup, Select the Voice Call Handler component.

The Voice Call Handler Configuration popup appears.

2. In the Voice Channel Type selection list, Select the Hoot-n-Holler option.

The parameters required for this channel type are activated on the popup.

3. In the HnH Action Definition entry field, create the hoot-n-holler "Action 0" by typing the action using the following format, then selecting the Add Action button:

ACTION0 BEGIN nnnnnnn XXXX

where nnnnnnn is the address/phone number you want this call to connect to.

and where XXXX is the voice call algorithm used for the call. For a description of these voice call algorithms, see Section 9.2.5.1 Voice Call Algorithms in Chapter 9 of Configuring and Monitoring a Network Exchange 2200.

The action appears in the Hoot-n-Holler Actions selection list.

4. In the HnH Action Definition entry field, create the hoot-n-holler "Action 1" by typing in the action using the following format, then selecting the Add Action button:

ACTION1 END

The action appears in the Hoot-n-Holler Actions selection list. Once you set up the required primitive, the call to the address designated in Action 0 ends when the Time Manager statistic you created in the previous section reaches a value of 0.

Note: You can create up to 10 hoot-n-holler actions. You only need to create one "END" action that can be used to terminate any active HnH call.

Creating a Primitive

The following steps show how to create a primitive that triggers the correct hoot-n-holler action to occur based on the value of the Time Manager statistic.

1. On the Top Level Voice/Fax Channel Info popup, Select the General Information button. On the General Information popup, Select the Primitive Information button.

The Status Configuration Information popup appears.

2. Create the RLX Action0 primitive by entering the following values:

In the Primitive Name field, type RLX Action0.
In the Dividend Statistic list, select Time of Day Statistic 1.
In the Divisor Statistic list, select Global Statistic 100.
Leave the thresholds and weights at their default values (minor 1 and major 10).
Under Transitions select Green -> Yellow.
Under Transition Actions select VCH Action 0.
Under Transitions select To Green.
Under Transition Actions select VCH Action 1.
Select the Add button.

The primitive name appears in the Status Primitives selection list. This primitive activates the hoot-n-holler action that you set up in the previous section.

3. To update the database and implement the configuration changes, Popdown to the General Information popup and Select the Update General Information button.

4. To update the database and implement the configuration changes in the Voice Call Handler, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

The channel launches a call to the address designated in the hoot-n-holler action at the time you designated in the Time Manager.

Configuring the Channel Driver Component

The Channel Driver component controls the transfer of signalling information and voice frames on individual voice channels.

Note: Channel Driver Parameters for Analog Voice describes each field on the Channel Driver Configuration popup. In addition, the online help text describes these fields.

Select the NXA Channel Driver component from the top level info popup of a voice channel.

The Channel Driver Configuration popup appears.

Notes:

On some NetrixView 2000 screens you may have difficulty viewing the lowest fields on the Channel Driver Configuration popup. If this occurs, you can resize the screen until it displays the full popup. Simply pull the right side of the window closer in, making the window narrower. Continue until the entire popup is visible.
Initially, you should leave most of these parameters at their default settings, adjusting them as necessary to improve system performance. However, there are certain parameters that you should set according to your network configuration. These parameters are listed below.

Setting Up the Channel Driver Component

1. Set the Electrical Protocol for the voice channels. Make sure this setting is compatible with the protocol used on the analog equipment to which this channel connects. (The equipment at the remote end of the call, however, is not required to use the same electrical protocol used at this end.)

2. Set the telephone Signaling Protocol to use on this channel.

3. In the Dial Method to the PBX field, specify how to transmit to the PBX, dialing digits received from the switch.

4. In the Dialing Method from the PBX field, specify how to handle dialing digits received from the PBX.

5. Set the Voice Compression Algorithm used for calls originating on this channel.

Note: For calls between two ports configured with different voice call algorithms, the algorithm on the initiating side of the call is used for the call.

You can control the amount of delay through the voice-compression engine. A higher delay causes more data to be packed into each frame, resulting in a lower frame-switching rate, without appreciably noticeable delay to the human ear.

If you use ACELP as the voice compression algorithm, you can control the delay using ACELP Additional Frame Delay.
If use Low Bit Rate ACELP, you can control the delay using the Low Bit Rate ACELP Additional Frame Delay.)

6. Use the Transmit and Receive TLP (Transmission Level Points) fields to control the quality and volume of the audio sent to and received from external voice equipment.

7. In the Echo Canceller Enable field, specify whether or not to run echo cancellation software on this channel. If the PBX supplies echo suppression, select disable.

8. In the Echo Suppression field, specify whether Silence or recorded Background Noise is played during echo suppression. On connections with high levels of background noise, you may wish to specify Background Noise. This helps avoid complete dropouts of far-end audio received during near-end speech.

9. In the Maximum Fax Rate field, set the maximum rate at which this channel processes faxes.

For shorter fax transmission time, select a higher rate.
To give voice and data traffic preference over faxes, select a lower rate.
To prohibit faxes on this line, use No FAX Allowed

10. To compensate for delay variances in the network, adjust the Maximum Jitter Compensation Delay.

11. If you plan to allow oversubscription of ACELP calls on SFTM trunks in the network, enable Voice Activity Detection. This feature takes advantage of the naturally-occurring silences in voice calls to enable a single 64 kbps trunk, for example, to carry up to 12 voice calls at one time.

12. To update the database and implement the configuration changes in the Channel Driver, Popdown to the Top Level Voice/Fax Channel Info popup and Select the Update & Config button.

Channel Driver Parameters for Analog Voice

This section describes the parameters you can set on the Channel Driver Configuration popup.

Electrical Protocol


E&M	E&M ("ear & mouth" or "earphone & microphone") leads are used to transfer off-hook and pulse dialing information during call setup. The analog voice module supports E&M interface types 1, 2, and 5. Each type differs in the voltage and currents used to assert the E&M leads, and in the grounding scheme used. Note: When using any of the E&M choices, you should use the Immediate Start or Wink Start Signaling Protocol. (You can also use Delay Dial, if required.) E&M Type 1—North American standard. E&M Type 2—This type is rarely used, and requires the use of two extra leads (signal ground and signal battery) to carry the reference ground and reference battery voltage to both ends of the local (PBX to XV) connection. E&M Type 5—Frequently used in Europe, the United Kingdom, and former British possessions.
FXO Ground Start	Foreign Exchange Office (FXO) FXO Ground Start: "Tip" and "Ring" refer to the two wires of the interface. The most common application is FXO Ground Start—Ring, wherein the protocol seizes the Ring lead to ground. For the UAI daughter card, select FXO Ground Start - RING. The UAI daughter card has the alternate capability of using the FXO Ground Start - TIP protocol—for example, if the wires have been reversed. Note: When using FXO Ground Start, you should use the Immediate Start Signaling Protocol.
FXO Loop Start	The analog voice module acts as a telephone, where The outbound signal is loop (the PBX detects loop by the flow of current). The inbound signal from the PBX is ring. This causes the analog voice module to detect ring voltage and close the loop, which stops the ring signal. Note: When using FXO Loop Start, you should use the Immediate Start Signaling Protocol.
FXS Loop Start	The analog voice module expects to connect to a telephone, fax machine, or keysystem. The outbound signal is ring (the phone/fax machine/keysystem detects ring voltage and closes the loop, which stops the ring signal). The inbound signal is loop (the analog voice module detects loop by the flow of current). Note: When using FXS Loop Start, use either the Immediate Start or Phone Signaling Protocol. If the channels connect to a telephone, use the Phone signalling protocol, which does not pass digits to the connected telephone equipment. other analog equipment (such as a PBX or keysystem), use the Immediate Start signalling protocol. You must also ensure that Immediate Off Hook is enabled.
None	This channel is not being used. Select None to ensure that the router does not monitor connection states for this channel.

Signaling Protocol

Sets the telephone signalling protocol to use on this channel.

Delay Dial

Timing for this protocol is similar to wink start timing, except that the E lead response to M lead assertion is not a pulse of fixed length. Instead, Delay Dial asserts the E lead immediately after seeing the M lead assertion, and the E lead remains asserted until dialing begins. The voice path is established once the local exchange of delay pulses takes place. (If no delay pulses are exchanged within 14 seconds, the voice path is automatically opened.)

Immediate Start

Provides a complete audio path to the far end during dialing, allowing you to hear call progress tones from the far end. You typically use this protocol on ports connecting to PBXs, keysystems, or central offices (unless the PBX or central office uses Wink Start). The PBX, keysystem, or central office generates the ringback tone to the call's originating end.

Hoot and Holler

Also known as Transparent, this option lets you allocate bandwidth for transparent connections during scheduled time periods. Transparent connections are useful for applications where a permanent connection without call setup time is desired.
During scheduled times, the analog voice module maintains a constant voice path and passes E&M leads transparently through the analog voice module.

Wink Start

When the trunk side wishes to seize the line, it asserts the M lead, then waits for a "wink" pulse on the E lead to acknowledge completion of the connection. Wink pulses range from 140 to 200 milliseconds in length. The voice path is established once the local exchange of wink pulses has taken place. (If no wink pulses are exchanged within 14 seconds, the voice path is automatically opened.)

Phone

Generates a ringing signal on the call's originating channel and does not complete the audio path until the far end is off hook. This protocol does not pass digits to the connected telephone equipment. Therefore, we recommend it for use with telephones, but not with other analog equipment, such as a PBX or keysystem. (This is a Nx Networks proprietary protocol.)

Delay Dial	Timing for this protocol is similar to wink start timing, except that the E lead response to M lead assertion is not a pulse of fixed length. Instead, Delay Dial asserts the E lead immediately after seeing the M lead assertion, and the E lead remains asserted until dialing begins. The voice path is established once the local exchange of delay pulses takes place. (If no delay pulses are exchanged within 14 seconds, the voice path is automatically opened.)
Immediate Start	Provides a complete audio path to the far end during dialing, allowing you to hear call progress tones from the far end. You typically use this protocol on ports connecting to PBXs, keysystems, or central offices (unless the PBX or central office uses Wink Start). The PBX, keysystem, or central office generates the ringback tone to the call's originating end.
Hoot and Holler	Also known as Transparent, this option lets you allocate bandwidth for transparent connections during scheduled time periods. Transparent connections are useful for applications where a permanent connection without call setup time is desired. During scheduled times, the analog voice module maintains a constant voice path and passes E&M leads transparently through the analog voice module.
Wink Start	When the trunk side wishes to seize the line, it asserts the M lead, then waits for a "wink" pulse on the E lead to acknowledge completion of the connection. Wink pulses range from 140 to 200 milliseconds in length. The voice path is established once the local exchange of wink pulses has taken place. (If no wink pulses are exchanged within 14 seconds, the voice path is automatically opened.)
Phone	Generates a ringing signal on the call's originating channel and does not complete the audio path until the far end is off hook. This protocol does not pass digits to the connected telephone equipment. Therefore, we recommend it for use with telephones, but not with other analog equipment, such as a PBX or keysystem. (This is a Nx Networks proprietary protocol.)

Normal	Does nothing.
Inverted	Changes the polarity
Set	Always shows M-lead present.
Clear	Always shows M-lead absent.

Enable

The analog voice module goes off-hook toward the originating PBX immediately after detecting off hook. Must be enabled for FXS

Disable

The analog voice module does not send an answering off hook signal until it receives a Call Answered signal from the remote end of the call.

Enable	The analog voice module goes off-hook toward the originating PBX immediately after detecting off hook. Must be enabled for FXS
Disable	The analog voice module does not send an answering off hook signal until it receives a Call Answered signal from the remote end of the call.

Flash Enable

Sets how the analog voice module detects and generates flashes.

Enable

The analog voice module interprets M-lead disassertion longer than the minimum flash detect time and shorter than the disconnect time as hook flashes and generates a flash message toward the switch when this occurs.
It also generates an E-lead disassertion of the flash generate time when it receives a flash message from the switch.

Disable

The analog voice module does not generate or detect flashes.

Enable	The analog voice module interprets M-lead disassertion longer than the minimum flash detect time and shorter than the disconnect time as hook flashes and generates a flash message toward the switch when this occurs. It also generates an E-lead disassertion of the flash generate time when it receives a flash message from the switch.
Disable	The analog voice module does not generate or detect flashes.

If disabled, CSA pulses are not generated or detected.

Specifies whether or not to generate outbound dial tone to a channel that has initiated a call. If enabled, dial tone is generated until the first inbound dialed digit is detected or until a Connect Accept is received from the switch. The primary use for this setting is to disable dial tone on Automatic Ring Down channels, which do not require dialed digits to place a call.

Dial Method to the PBX

Specifies how to transmit to the PBX dialing digits received from the switch. Digits arrive from the switch with an indication of whether they were generated as pulse or tone digits.

Specifies how to handle dialing digits the analog voice module receives from the switch. Digits arrive from the switch with an indication of whether they were generated as pulse or tone digits. You can override that indication using this parameter. A setting of transparent tells the analog voice module to reproduce the digits as the switch generated them.

Disabled

Disable dialing to the PBX. The analog voice module does not generate or transmit dialed digits.

Pulse Only

Overrides the indication set on incoming digits to pulse digits.

Tone Only

Overrides the indication set on incoming digits to tone digits.

Transparent

Reproduces the digits as they were entered, pulse or tone.

Disabled	Disable dialing to the PBX. The analog voice module does not generate or transmit dialed digits.
Pulse Only	Overrides the indication set on incoming digits to pulse digits.
Tone Only	Overrides the indication set on incoming digits to tone digits.
Transparent	Reproduces the digits as they were entered, pulse or tone.

Both	Accept both tone and pulse digits.
Tone Only	Ignore pulse digits.
Disabled	Ignore all dialed digits.
Pulse Only	Ignore tone digits.

You can set up a voice channel to use a voice call algorithm for transmission of calls via trunks connected to other 3000 Series voice modules or to Network Exchange 2200 voice equipment. Uncompressed voice channels use 64 kbps of bandwidth; voice compression allows information to be packed into smaller bandwidth.

When the analog voice module receives a voice call on a channel configured for voice compression, it compresses the voice signals according to the configured algorithm, and switches them through the network. The analog voice module on the other end of the call decompresses the voice signals before passing them to the voice subscriber at its end of the call.

Note: For calls between two ports configured with different voice call algorithms, the algorithm configured on the initiating side of the call is used for the call, and overrides the algorithm configured on the receiving side of the call.

For detailed information on voice compression and voice call algorithms, see Section 9.2.5 Voice Compression in Chapter 9 of the Configuring and Monitoring a Network Exchange 2200 guide.

TDHS

Time Domain Harmonic Scaling (TDHS) detects and compresses pitch periods (digitized samples of speech). Also includes linear prediction coding (LPC) to further compress the amount of information needed to transmit and recreate the signal. (Actual throughput bandwidth levels vary between 9 kbps and 12 kbps, depending on break factor configuration and system load.) Calls using the TDHS voice call algorithm are assigned the TDHS class of service.

4.8Kbps CELP
OR
7.4Kbps CELP

CELP (Code Excited Linear Prediction) breaks a fixed-size sampled-input signal into blocks of samples that are processed according to standard models. The speech signal's format structure and pitch are modeled and quantified for recreation at the remote end.
You can set the bandwidth of the compressed voice stream to 4.8 kbps or 7.4kbps. Calls using either of these settings are assigned the CELP class of service.

4.8 kbps CELP—Allocates a CIR of 6.2 kbps (this includes the overhead required by the 2200 network).
7.4 kbps CELP—Provides a voice quality that is slightly higher than 4.8 kbps. Allocates a CIR of 9 kbps (this includes the overhead required by the 2200 network).

ACELP
OR
Low Bit Rate ACELP

ACELP (Algebraic Code Excited Linear Prediction) is the highest quality compressed voice call algorithm.
ACELP is a modified CELP algorithm that supports silence suppression (also known as voice activity detection, or VAD), so that trunk bandwidth is used only when active speech is present. This permits more efficient use of network bandwidth by allowing voice or data calls to make use of the bandwidth that naturally occurring silences in voice calls provide.
You can use either of the following ACELP settings.

Low Bit Rate ACELP—the peak bandwidth of the compressed voice stream is 5.5 kbps. Allocates a CIR of 7.4 kbps (this includes the overhead required by the 2200 network).
ACELP—the peak bandwidth of the compressed voice stream is 8 kbps. Allocates a CIR of 9.8 kbps (this includes the overhead required by the 2200 network).
Calls using the ACELP or Low Bit Rate ACELP voice call algorithm are assigned the CELP class of service.

H.323 (PCM)

TDHS	Time Domain Harmonic Scaling (TDHS) detects and compresses pitch periods (digitized samples of speech). Also includes linear prediction coding (LPC) to further compress the amount of information needed to transmit and recreate the signal. (Actual throughput bandwidth levels vary between 9 kbps and 12 kbps, depending on break factor configuration and system load.) Calls using the TDHS voice call algorithm are assigned the TDHS class of service.
4.8Kbps CELP OR 7.4Kbps CELP	CELP (Code Excited Linear Prediction) breaks a fixed-size sampled-input signal into blocks of samples that are processed according to standard models. The speech signal's format structure and pitch are modeled and quantified for recreation at the remote end. You can set the bandwidth of the compressed voice stream to 4.8 kbps or 7.4kbps. Calls using either of these settings are assigned the CELP class of service. 4.8 kbps CELP—Allocates a CIR of 6.2 kbps (this includes the overhead required by the 2200 network). 7.4 kbps CELP—Provides a voice quality that is slightly higher than 4.8 kbps. Allocates a CIR of 9 kbps (this includes the overhead required by the 2200 network).
ACELP OR Low Bit Rate ACELP	ACELP (Algebraic Code Excited Linear Prediction) is the highest quality compressed voice call algorithm. ACELP is a modified CELP algorithm that supports silence suppression (also known as voice activity detection, or VAD), so that trunk bandwidth is used only when active speech is present. This permits more efficient use of network bandwidth by allowing voice or data calls to make use of the bandwidth that naturally occurring silences in voice calls provide. You can use either of the following ACELP settings. Low Bit Rate ACELP—the peak bandwidth of the compressed voice stream is 5.5 kbps. Allocates a CIR of 7.4 kbps (this includes the overhead required by the 2200 network). ACELP—the peak bandwidth of the compressed voice stream is 8 kbps. Allocates a CIR of 9.8 kbps (this includes the overhead required by the 2200 network). Calls using the ACELP or Low Bit Rate ACELP voice call algorithm are assigned the CELP class of service.
H.323 (PCM)

Transmit and Receive TLP

Transmission Level Points (TLPs) set the amplitude of the audio signal sent to or received from the voice equipment. A general guideline for adjusting Transmit and Receive TLP levels is Transmit affects volume and Receive affects quality.

Note: The default values (-5 Transmit, 0 Receive) are considered reasonable starting points when configuring TLP levels on voice channels.

When you set the TLPs, your goal is to bring the actual voice signal level to 0 dBm for travel across the network. In practical terms, this means you should set the TLPs equal to the dBm setting for the voice equipment's receive level. Positive numbers represent gain, negative numbers represent attentuation.

Figure 1 shows example Transmit and Receive TLP settings.

Figure 1 Transmit and Receive TLP

In this example,

the voice signal coming from the voice equipment to 3000 Series A is -12 dBm. The Receive TLP is set to -12 dBm, which causes the analog channel to expect to receive voice signals with an average range of -12 dBm, and correct accordingly to ensure that voice signals travel across the network at 0 dBm. (This means that when you set the Receive TLP to a negative value, the analog voice module applies gain to the signal.)
the voice equipment attached to 3000 Series B expects a voice signal of -16 dBm. The Transmit TLP is set to -16 dBm. This setting causes the analog voice module, which expects to receive voice signals from the network at 0 dBm, to attentuate the signal 16 dBm to send them to the voice equipment at -16 dBm.

Note: Because the receive TLP is specified as the measured receive level that requires correction, negative numbers represent the gain that must be applied to bring a low value up to zero, while positive numbers represent the attentuation that must be applied to bring a high value down to zero.

Troubleshooting Voice Quality and Volume

If the Transmit TLP is set with a value that is too low, the user will report that the voice is too quiet. If the Transmit TLP is set with a value that is too high, voice distortion results.

It is important to remember that incorrect settings at one end of the call may result in symptoms that seem to point to incorrect settings at the other end. In some cases, slight (1 to 2 dBm) adjustments are necessary to optimize voice quality or add small amounts of gain or attenuation to suit the users' preferences. In this case, you should first ensure that the Transmit and Receive TLPs on both sides of the call are set correctly to correspond with the requirements of the connected phone equipment. Then you can make adjustments as follows:

If the user hears . . . then . . .

good quality, but at a low level
increase the Transmit TLP on the local analog voice module.

good quality, but at a high level
reduce the Transmit TLP on the local analog voice module.

poor quality at a low level
increase the Receive TLP on the remote analog voice module.

poor quality at a high level
increase the Receive TLP on the remote analog voice module.

poor quality at an appropriate level
reduce the Receive TLP at the remote analog voice module, and reduce the Transmit TLP at the local analog voice module.

If the user hears . . .	then . . .
good quality, but at a low level	increase the Transmit TLP on the local analog voice module.
good quality, but at a high level	reduce the Transmit TLP on the local analog voice module.
poor quality at a low level	increase the Receive TLP on the remote analog voice module.
poor quality at a high level	increase the Receive TLP on the remote analog voice module.
poor quality at an appropriate level	reduce the Receive TLP at the remote analog voice module, and reduce the Transmit TLP at the local analog voice module.