Cisco Voice Application

Transcription

1 Titlepage Supports Management Module SM-CIS1006 Device Management

2 Copyright Notice Document Copyright 2002-present by Aprisma Management Technologies, Inc. All rights reserved worldwide. Use, duplication, or disclosure by the United States government is subject to the restrictions set forth in DFARS (c)(1)(ii) and FAR Liability Disclaimer Aprisma Management Technologies, Inc. ( Aprisma ) reserves the right to make changes in specifications and other information contained in this document without prior notice. In all cases, the reader should contact Aprisma to inquire if any changes have been made. The hardware, firmware, or software described in this manual is subject to change without notice. IN NO EVENT SHALL APRISMA, ITS EMPLOYEES, OFFICERS, DIRECTORS, AGENTS, OR AFFILIATES BE LIABLE FOR ANY INCIDENTAL, INDIRECT, SPECIAL, OR CONSEQUENTIAL DAMAGES WHATSOEVER (INCLUDING BUT NOT LIMITED TO LOST PROFITS) ARISING OUT OF OR RELATED TO THIS MANUAL OR THE INFORMATION CONTAINED IN IT, EVEN IF APRISMA HAS BEEN ADVISED OF, HAS KNOWN, OR SHOULD HAVE KNOWN, THE POSSIBILITY OF SUCH DAMAGES. Trademark, Service Mark, and Logo Information SPECTRUM, IMT, and the SPECTRUM IMT/VNM logo are registered trademarks of Aprisma Management Technologies, Inc., or its affiliates. APRISMA, APRISMA MANAGEMENT TECHNOLOGIES, the APRISMA MANAGEMENT TECHNOLOGIES logo, MANAGE WHAT MATTERS, DCM, VNM, SpectroGRAPH, SpectroSERVER, Inductive Modeling Technology, Device Communications Manager, SPECTRUM Security Manager, and Virtual Network Machine are unregistered trademarks of Aprisma Management Technologies, Inc., or its affiliates. For a complete list of Aprisma trademarks, service marks, and trade names, go to All referenced trademarks, service marks, and trade names identified in this document, whether registered or unregistered, are the intellectual property of their respective owners. No rights are granted by Aprisma Management Technologies, Inc., to use such marks, whether by implication, estoppel, or otherwise. If you have comments or concerns about trademark or copyright references, please send an to spectrum-docs@aprisma.com; we will do our best to help. Restricted Rights Notice (Applicable to licenses to the United States government only.) This software and/or user documentation is/are provided with RESTRICTED AND LIMITED RIGHTS. Use, duplication, or disclosure by the government is subject to restrictions as set forth in FAR (June 1987) Alternate III(g)(3) (June 1987), FAR (June 1987), or DFARS (c)(1)(ii) (June 1988), and/or in similar or successor clauses in the FAR or DFARS, or in the DOD or NASA FAR Supplement, as applicable. Contractor/manufacturer is Aprisma Management Technologies, Inc. In the event the government seeks to obtain the software pursuant to standard commercial practice, this software agreement, instead of the noted regulatory clauses, shall control the terms of the government's license. Virus Disclaimer Aprisma makes no representations or warranties to the effect that the licensed software is virus-free. Aprisma has tested its software with current virus-checking technologies. However, because no antivirus system is 100 percent effective, we strongly recommend that you write-protect the licensed software and verify (with an antivirus system in which you have confidence) that the licensed software, prior to installation, is virus-free. Contact Information Aprisma Management Technologies, Inc. 273 Corporate Drive Portsmouth, NH Phone: U.S. toll-free: Web site: Device Management Page 2

3 Contents INTRODUCTION 5 Purpose and Scope...5 Required Reading...5 Supported Devices...6 The SPECTRUM Model...6 A Cisco Voice Model...8 TASKS 9 EVENTS 10 CISCO VOICE APPLICATION Voice Analog Interface...12 CiscoVoiceApp Voice Analog Interface General Information View...12 Configuration...12 Status...13 CiscoVoiceApp Voice Analog Interface Ear and Mouth Information View...15 Config...15 Status...16 Timing...17 CiscoVoiceApp Voice Analog Interface Foreign Exchange Office Information View...18 Configuration Table...18 Status Table...19 Timing Table...19 CiscoVoiceApp Voice Analog Interface Foreign Exchange Station Information View...20 Configuration Table...20 Status Table...20 Timing Table...21 CiscoVoiceApp Voice Analog Interface Configuration Information View...22 RFC2128 Dial Control...24 RFC2128 Dial Control Peer Information View...25 Peer Configuration...25 Peer Status RFC2128 Dial Control Active Calls Monitor View..29 Active Calls Table RFC2128 Dial Control Call History Table View...32 Call History Table...32 Cisco Dial Control Call History Table View...34 Cisco Dial Control Call History Table...34 Cisco Voice Dial Control CiscoVoiceApp General Peer Configuration Table View General Peer Configuration...37 CiscoVoiceApp Specific Peer Configuration Table View 38 Voice Specific Peer Configuration Device Management Page 3

4 Contents Contents CiscoVoiceApp VoIP Specific Peer Configuration Table View...39 VoIP Specific Peer Configuration Table...39 CiscoVoiceApp Gateway Call Active Table View...42 Gateway Call Active Table...42 CiscoVoiceApp Gateway VoIP Call Active Table View 43 Voice over IP Call Active Table...44 CiscoVoiceApp Gateway Call History Table View.46 Voice Call History Table...46 CiscoVoiceApp Gateway VoIP Call History Table View 47 Voice over IP Call History Table...47 MODEL INFORMATION VIEW 49 INDEX 50 Device Management Page 4

5 Introduction This section introduces the SPECTRUM Device Management documentation for the. This introduction contains the following topics: Purpose and Scope Required Reading Supported Devices (Page 6) The SPECTRUM Model (Page 6) A Cisco Voice Model (Page 8) Purpose and Scope Use this document as a guide for managing the with SPECTRUM management module SM-CIS1006. This document describes the icons, menus, and views that enable you to remotely monitor, configure, and troubleshoot the Cisco Voice application through software models in your SPECTRUM database. Information specific to SM-CIS1006 is what is primarily included in this document. For general information about device management using SPECTRUM and explanations of SPECTRUM functionality and navigation techniques, refer to the topics listed under Required Reading. Required Reading To use this documentation effectively, you must be familiar with the information covered by the other SPECTRUM online documents listed below. Getting Started with SPECTRUM for Operators Getting Started with SPECTRUM for Administrators How to Manage Your Network with SPECTRUM SPECTRUM Views SPECTRUM Menus SPECTRUM Icons SPECTRUM Software Release Notice Device Management Page 5

6 Introduction Supported Devices SPECTRUM management module SM-CIS1006 currently lets you model the Cisco Voice Application. This application provides an integrated voice and data management solution. The following devices are supported: Cisco Router - with Firmware 11.3T and up Cisco MC with Firmware 11.3T and up Cisco Access Servers - with Firmware 11.3T and up The SPECTRUM Model Supported Devices The model type for the is CiscoVoiceApp. The application-specific Icon Subviews menu options available from the CiscoVoiceApp Application icon are listed in the table on Page 7. Device Management Page 6

7 Introduction The SPECTRUM Model Option Accesses the... Voice Analog Interface Subviews below: - General Info CiscoVoiceApp Voice Analog Interface General Information View (Page 12) - Ear and Mouth CiscoVoiceApp Voice Analog Interface Ear and Mouth Information View (Page 15) - Foreign Exchange Office Info CiscoVoiceApp Voice Analog Interface Foreign Exchange Office Information View (Page 18) - Foreign Exchange Station Info CiscoVoiceApp Voice Analog Interface Foreign Exchange Station Information View (Page 20) Voice IF Configuration CiscoVoiceApp Voice Analog Interface Configuration Information View (Page 22) RFC 2128 Dial Control Subviews below: - Peer Information RFC2128 Dial Control Peer Information View (Page 25) - Call Active RFC2128 Dial Control Active Calls Monitor View (Page 29) - Call History RFC2128 Dial Control Call History Table View (Page 32) Cisco Dial Control Call History Cisco Dial Control Call History Table View (Page 34) Cisco Voice Dial Control Subviews below: - General Peer Configuration CiscoVoiceApp General Peer Configuration Table View (Page 36) - Specific Peer Configuration CiscoVoiceApp Specific Peer Configuration Table View (Page 38) - VoIP Specific Peer Configuration CiscoVoiceApp VoIP Specific Peer Configuration Table View (Page 39) - Gateway Call Active CiscoVoiceApp Gateway Call Active Table View (Page 42) - Gateway VoIP Call Active CiscoVoiceApp Gateway VoIP Call Active Table View (Page 43) - Gateway Call History CiscoVoiceApp Gateway Call History Table View (Page 46) - Gateway VoIP Call History CiscoVoiceApp Gateway VoIP Call History Table View (Page 47) Model Information Model Information View (Page 49) Device Management Page 7

8 Introduction A Cisco Voice Model The Cisco Voice application lets the user place telephone calls and send faxes over IP-based data networks with an exceptional Quality of Service (QOS) and cost benefit. In addition, voice messages can be prepared using a telephone and delivered to an integrated voice/data mailbox using internet or intranet services. Figure 1 shows an example of a network using Cisco Voice over IP. A Cisco Voice Model The rest of this document covering management module SM-CIS1006 is organized as follows: Tasks (Page 9) Events (Page 10) (Page 11) Model Information View (Page 49) Figure 1: Example of a Cisco Voice over IP Network I Voice Port Voice Port Voice Port Voice Port Voice Port Voice Port Cisco Router IP Cloud Cisco Router Voice Port Device Management Page 8

9 Tasks This section contains an alphabetical list of device management tasks, with each task providing one or more links to views that let you perform the task. Administrative Information (check) Model Information View (Page 49) Dial Digits Prefix (set) Voice Specific Peer Configuration (Page 38) Encapsulation Type (set) General Peer Configuration (Page 37) Event Messages (view/create) Events (Page 10) Global Call ID (check) Gateway Call Active Table (Page 42) Voice over IP Call Active Table (Page 44) Voice Call History Table (Page 46) Voice over IP Call History Table (Page 47) Incoming Call Security Level (set) RFC2128 Dial Control Peer Information View (Page 25) Information Transfer Services Type (check) Status (Page 13) Peer Call Traps (enable/disable) RFC2128 Dial Control Peer Information View (Page 25) Session Target (set) Voice Specific Peer Configuration (Page 38) VoIP Specific Peer Configuration Table (Page 39) Terminating Impedance (check/configure) CiscoVoiceApp Voice Analog Interface General Information View (Page 12) VoIP Counter (reset) CiscoVoiceApp Gateway VoIP Call Active Table View (Page 43) Device Management Page 9

10 Events This section lists the Events that are available for the Cisco Voice application. The Cisco Voice application has several event messages that are specific to its devices. If these messages are not sufficient and you wish to create your own messages, or view the existing messages for the Cisco Voice, you can do so using the ECEditor. The event messages for the Cisco Voice, which include Event , Event , and Event can be found in the following directory: ~/SG-Support/CsEvFormat Device Management Page 10

11 This section provides a description of the Cisco Voice application and its associated application-specific subviews available in SPECTRUM. This major application (CiscoVoiceApp) provides access to the following application-specific views: CiscoVoiceApp Voice Analog Interface General Information View (Page 12) CiscoVoiceApp Voice Analog Interface Ear and Mouth Information View (Page 15) CiscoVoiceApp Voice Analog Interface Foreign Exchange Office Information View (Page 18) CiscoVoiceApp Voice Analog Interface Foreign Exchange Station Information View (Page 20) CiscoVoiceApp Voice Analog Interface Configuration Information View (Page 22) RFC2128 Dial Control Peer Information View (Page 25) RFC2128 Dial Control Active Calls Monitor View (Page 29) RFC2128 Dial Control Call History Table View (Page 32) Cisco Dial Control Call History Table View (Page 34) CiscoVoiceApp General Peer Configuration Table View (Page 36) CiscoVoiceApp Specific Peer Configuration Table View (Page 38) CiscoVoiceApp VoIP Specific Peer Configuration Table View (Page 39) CiscoVoiceApp Gateway Call Active Table View (Page 42) CiscoVoiceApp Gateway VoIP Call Active Table View (Page 43) CiscoVoiceApp Gateway Call History Table View (Page 46) CiscoVoiceApp Gateway VoIP Call History Table View (Page 47) Device Management Page 11

12 Voice Analog Interface Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice Analog Interface and one of the four subviews. This selection accesses the following views, described below: CiscoVoiceApp Voice Analog Interface General Information View CiscoVoiceApp Voice Analog Interface Ear and Mouth Information View (Page 15) CiscoVoiceApp Voice Analog Interface Foreign Exchange Office Information View (Page 18) CiscoVoiceApp Voice Analog Interface Foreign Exchange Station Information View (Page 20) CiscoVoiceApp Voice Analog Interface General Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice Analog Interface > General Info. This selection opens the Voice Analog Interface General Information View and shows the following information. Configuration The configuration table contains the following standard configuration information of the Analog Telephony Interface. Double-clicking an entry in this table opens an Analog Interface Configuration view for that entry where you can configure that entry. IF An entry in the Configuration Table for each voice analog interface. The entry is created when the voice analog hardware is detected. Impedance Specifies the terminating impedance of voice analog interfaces. Integrated DSP Indicates whether the interface has an integrated Digital Signal Processing (DSP) unit. Device Management Page 12

13 Status The status table contains the following general information about the status of Analog Telephony Interface including the error statistics. IF An entry in the status table for each Analog voice interface. Info Type Table 1 lists the information transfer services for the current or last call. Table 1: Type Information Type Description none voice g3fax No information transfer service Analog voice Group 3 Fax Device Management Page 13

14 Maintenance Mode Table 2 lists the maintenance modes of the Analog voice interface. Table 2: Mode Maintenance Modes (Continued) Definition none Mode Table 2: hostcompressed Loopback hostuncompressed Loopback Maintenance Modes Definition The interface is not in maintenance mode. This mode is set by the managed system to perform the compressed voice loopback test. The voice packet is passed by voice application to the interface associated CODEC. Instead of feeding these into the CODEC for decompression, they are immediately sent back to the voice application as if they had originated locally and been compressed. This mode is set by the managed system to perform the uncompressed voice loopback test. The voice packet is sent by voice application all the way through the interface associated CODEC and then turned around instead of being sent to the telephony endpoint. ifcompressedloop back ifuncompressed Loopback This mode is set by the managed system to perform the interface uncompressed voice loopback test for the external test equipment that is connected to the voice interface. The voice samples are sent from the test equipment all the way through the interface associated CODEC and then turned around instead of being sent to voice application. This mode is set by the managed system to perform the interface uncompressed voice loopback test for the external test equipment that is connected to the voice interface. The voice samples are sent from the test equipment to the interface associated CODEC. Instead of feeding the voice samples into the CODEC for compression, they are immediately sent back to the telephony interface. Status Signal Errors Accumulated number of signaling protocol errors that are detected in the interface since system startup. Device Management Page 14

15 CiscoVoiceApp Voice Analog Interface Ear and Mouth Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice Analog Interface > Ear and Mouth Info. This view provides the following tables: Config Status (Page 16) Timing (Page 17) Config The configuration table contains the current settings of Ear and Mouth configurable parameters, described below. IF An entry in the standard configuration table for each E&M (Ear & Mouth) tie trunk interface. Signal Type The signaling type of E&M tie trunk interface. The default value of this field is winkstart. Table 3 lists the available Signal Types. Type winkstart immediatedial delaydial Table 3: Signal Types Definition The calling side seized the line by going off-hook on its E lead but waits for a short off-hook wink indication on its M lead from the called side before sending address information as DTMF digits. The calling side seizes the line by going off-hook on its E lead and sends address information as DTMF digits. The calling side seizes the line by going off-hook on its E lead. After a timing interval, the calling side looks at the supervision from the called side. If a supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise, the calling side waits until the called side goes on-hook and then starts sending address information Device Management Page 15

16 Operation Specifies the operation of the E&M signal. Choices are: twowires = 2-wire operation fourwires = 4-wire operation The default value of this field is twowires. Type Specifies the E&M Lead Signaling type. Table 4 lists the available Signaling Types. Type Table 4: V -48V Signaling Type Definition /\/\/ detector/resister X normally open relay contact + normally closed relay contact G SB SG ground Signal Battery Signal Ground Dial Type Specifies the out dialing type of E&M interface dtmf = DTMF digit pulse = pulse The default value of this field is dtmf. Status The Ear and Mouth (E&M) interface table contains the following information about the status of the E&M tie trunk interface. IF An entry in the status table for each E&M interface. Incoming Seizure State This field contains the incoming seizure state of the E&M interface. If the value is true, then the interface has an incoming trunk seizure active; otherwise, it does not. Outgoing Seizure State This field contains the outgoing seizure state of the E&M interface. If the value is true, then the interface has an outgoing trunk seizure active; otherwise, it does not. Device Management Page 16

17 Timing The Ear and Mouth (E&M) tie trunk timing parameters table contains the configuration of the various E&M signaling protocol timing parameters, described below. IF An entry in the E&M Timing Parameters Table for each E&M interface. Digit Duration Outgoing DTMF digit duration in milliseconds. The default value of this field is 100 milliseconds. Inter-digit Duration Outgoing DTMF inter-digit duration in milliseconds. The default value of this field is 100 milliseconds. Pulse Rate Outgoing pulse dialing rate in pulses per second (pps). The default value of this field is 10 pps. Pulse Inter-digit Duration Outgoing Pulse dialing inter-digit timing in milliseconds. The default value of this field is 500 milliseconds. Clear Wait Duration Time of Inactive seizure signal to declare call cleared. The default value of this field is 400 milliseconds. Max Wink Wait Duration Maximum time to wait from the outgoing seizure of the line to the time of sending out a wink-pulse. The default value of this field is 200 milliseconds. Max Wink Duration Maximum outgoing wink duration (between winkpulse leading edge and trailing edge) for wink start signaling. The default value of this field is 200 milliseconds. Delay Start Minimum time to wait from outgoing seizure to out-dialing digits for delay start signaling. The default value of this field is 300 milliseconds. Max Delay Duration Maximum time to wait from outgoing seizure to out-dialing digits. If the far end does not allow to send digits after the duration specified in the field, a reorder tone will be sent to the calling party. The default value of this field is 2000 milliseconds. Device Management Page 17

18 Min Delay Pulse Width Minimum delay-dial pulse duration for incoming call. If the value of this field is 0, then no delaydial is sent to the far end for incoming call. The default value of this field is 140 milliseconds. CiscoVoiceApp Voice Analog Interface Foreign Exchange Office Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice Analog Interface > Foreign Exchange Office Info. This view provides three Foreign Exchange Office (FXO) tables: a Configuration Table, a Status Table (Page 19), and a Timing Table (Page 19). Configuration Table The Foreign Exchange Office (FXO) standard configuration table contains Foreign Exchange Office configuration parameters, one entry per FXO interface. The configuration information of this table is set by hardware with its default setting or CLI during the hardware installation time. Signal Type The signaling type of FXO interface. fxoloopstart = loop start fxogroundstart = ground start The default value of this field is fxoloopstart. Number of Rings The number of rings detected before closing loop. The default value of this field is 1 ring. Supervisory Disconnect Specifies whether a supervisory disconnect signal (CPC) will be presented to the interface when the connected party at the interface hangs up. The default value of this field is true. Dial Type Specifies the out dialing type of FXO interface. dtmf = DTMF digit pulse = pulse The default value of this field is dtmf. IF An entry in the standard configuration table for each FXO (Foreign Exchange Office) interface. Device Management Page 18

19 Status Table The Foreign Exchange Office (FXO) interface status table contains the following information about the various states of the interface. IF An entry in the status table for each FXO interface. Hook Status This field contains the hook status of FXO interface. Possible values are: onhook = Present the onhook to the connected equipment. offhook = Present the offhook to the connected equipment. Ring Detect This field contains the ring detect indication. If the value is true, then the interface detects ring from PBX or CO; otherwise, it does not. Ring Ground This field contains the incoming ring ground indication. It is only valid when the Signal Type is fxogroundstart. If the field is true, then the interface is presenting a ring ground to the connected equipment; otherwise, it does not. Tip Ground This field contains the incoming tip ground indication. It is only valid when the Signal Type is fxogroundstart. If the field is true, then the interface is detecting a tip ground at the interface; otherwise, it does not. Timing Table The Foreign Exchange Office (FXO) interface timing parameters table contains the configuration of the various FXO signaling protocol timing parameters. IF An entry in the FXO Timing Parameters Table for each FXO interface. Digit Duration Outgoing DTMF digit duration in milliseconds. The default value of this field is 100 milliseconds. Inter-Digit Duration Outgoing DTMF inter-digit duration in milliseconds. The default value of this field is 100 milliseconds. Pulse Rate Outgoing pulse dialing rate in pulses per second (pps). The default value of this field is 10 pps. Device Management Page 19

20 Pulse Inter-Digit Duration Outgoing pulse dialing inter-digit timing in milliseconds. The default value of this field is 500 milliseconds. CiscoVoiceApp Voice Analog Interface Foreign Exchange Station Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice Analog Interface > Foreign Exchange Station Info. This view provides three Foreign Exchange Station (FXS) tables: a Configuration Table, a Status Table, and a Timing Table (Page 21). Configuration Table This table contains the Foreign Exchange Station (FXS) configuration parameters, with one entry per FXS interface. The configuration information of this table is set by hardware with its default setting or CLI during hardware installation. Double-clicking an entry in this table opens an FXS IF Configuration view where you can set Signal Type and Ring Frequency for that entry. IF An entry in the standard configuration table for each FXS interface. Signal Type The signaling type of FXS interface. fxsloopstart = loop start fxsgroundstart = ground start The default value of this field is fxsloopstart. Ring Frequency The ring frequency to be used in the FXS interface. Possible values appear in Table 5. Table 5: FXS Interface Ring Frequency Value ringfrequency25 ringfrequency50 ringfrequency20 Ring Frequency 25 Hertz 50 Hertz 20 Hertz Status Table This table contains the hook status, ring active, and other statuses of the interface. IF An entry in the status table for each FXS interface. Device Management Page 20

21 Hook Status This field contains the hook status of FXS interface. onhook = the connected equipment (e.g.,phone) is onhook offhook = the connected equipment is off- Hook. Ring Detect This field contains the outgoing ring active indication. If the field value is true, then the interface is currently generating ring out to the Key Telephone System (KTS) or Plain Old Telephone (POT); otherwise, it does not. Ring Ground This field contains the incoming ring ground indication. It is only valid when the Signal Type is fxsgroundstart. If the field is true, then the interface is presenting a ring ground to the connected equipment; otherwise it does not. Tip Ground This field contains the incoming tip ground indication. It is only valid when the Signal Type is fxsgroundstart. If the field is true, then the interface is detecting a tip ground at the interface; otherwise, it does not. Timing Table This table contains the configuration of the various Foreign Exchange Station (FXS) signaling protocol timing parameters. IF An entry in the FXS Timing Table for each FXS interface. Digit Duration Outgoing DTMF digit duration in milliseconds. The default value of this field is 100 milliseconds. Inter-digit Duration Outgoing DTMF inter-digit duration in milliseconds. The default value of this field is 100 milliseconds. Device Management Page 21

22 CiscoVoiceApp Voice Analog Interface Configuration Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Voice IF Configuration. This view provides the Voice Interface Configuration table which contains the following information. Double-clicking an entry in this table opens a CiscoVoiceApp Voice Analog Interface Configuration Information view for that entry where you can make configuration changes to that entry. IF An entry in the Configuration Table for each voice analog interface or voice feature associated ISDN signaling channel. The entry is automatically created when voice analog interface hardware is found or ISDN signaling channel is configured and the ISDN associated voice support hardware is detected by the managed system. For the ISDN signaling channel associated entry, the entry is deleted after ISDN signaling channel is removed. Background Noise Enable The field indicates whether or not the background noise should be played to fill silence gaps if VAD is activated. The default value of this field is enabled. Non-Linear Processing Enable The field indicates whether or not the NOnlinear Processing is enabled for the interface. The default value of this field is enabled. Music on Hold Threshold The field specifies the Music On Hold Threshold for the interface. The default value of this field is -38 dbm. Input Gain This field contains the amount of gain inserted at the receiver side of the interface. The input gain settings only define a gain/loss relative to the 0 db setting. The absolute loss at the 0 db setting could be implementation dependent based on the desired network loss plan. The default value of this field is 0 db (decibel). Output Gain This field contains the amount of attenuation inserted at the transmit side of the interface. The output attenuation settings only define a loss relative to the 0 db setting. The absolute loss at the 0 db setting could be implementation dependent based on the desired network loss plan. The default value of this field is 0 db (decibel). Device Management Page 22

23 Echo Cancellation Enable The field specifies whether or not the Echo Cancellation is enabled for the interface. The default value of this field is enabled. Echo Canceller Coverage The field specifies the Echo Canceller coverage for the interface. This field is valid if the Echo Cancel Enable field is true. Choices are: echocanceller16ms - 16 milliseconds echo canceller coverage. echocanceller24ms - 24 milliseconds echo canceller coverage. echocanceller32ms - 32 milliseconds echo canceller coverage. The default value of this field is echocanceller16ms. Connection Mode The field indicates the connection mode of the interface. Table 6 lists the Connection Interface Mode. Table 6: Connection Mode normal trunk plar Connection Interface Mode Description The interface acts normally to handle incoming and outgoing calls. The interface acts in trunking mode. Trunking mode causes two interfaces (on either the same of different routers) to be always conferenced together, and for telephony signaling to be passed transparently through the VoIP system rather than trapped and interpreted by the session application on the router(s). The interface acts in PLAR (Private Line Automatic Ringdown) mode. PLAR is handled by associating a peer directly with an interface such that when the interface goes off-hook, that peer is used to setup the second call leg and conference them together without the caller having to dial any digits. Device Management Page 23

24 Connection Number This is the full E.164 phone number that is used to establish connection with trunking mode or PLAR mode. If the Connection Mode is normal, then this field will be a zero length string. Initial Digit Timeout This is the initial digit timeout that indicates the amount of time the managed system waits for an initial input digit from the caller. The timer is activated when the call is accepted and is deactivated upon digit input. If the timer expires, the caller is signaled through the appropriate tone and the call is abandoned. The value of zero disables the timer. The default value of this field is 10 seconds. Inter-digit Timeout This is the inter-digit timeout that indicates the amount of time the managed system waits for a subsequent input digit from the caller. The timer is started upon receipt of an input digit and restarted as each digit is received until the destination address is identified. If the timer expires and no destination address is identified, the caller is signaled through the appropriate tone and the call is abandoned. The value of zero disables the timer. The default value of this field is 10 seconds. Regional Tone This field contains the current regional setting for the tone, ring parameters to be used by this interface. RFC2128 Dial Control Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select RFC2128 Dial Control and one of the three subviews. This selection accesses the following views, described below: RFC2128 Dial Control Peer Information View (Page 25) RFC2128 Dial Control Active Calls Monitor View (Page 29) RFC2128 Dial Control Call History Table View (Page 32) Device Management Page 24

25 RFC2128 Dial Control Peer Information View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select RFC2128 Dial Control > Peer Information. This view contains Accept Mode and Trap Enable buttons as well as Peer Configuration and Peer Status (Page 28) tables, all described below. Accept Mode This button shows the security level for acceptance of incoming calls. Table 7 lists Accept Modes and their definitions. Security Level acceptnone acceptall acceptknown Table 7: Accept Modes Description Incoming calls will not be accepted. Incoming calls will be accepted, even if there is no matching entry in the Dial Control Peer Configuration Table. Incoming calls will be accepted only if there is a matching entry in the Dial Control Peer Configuration Table. Trap Enable This field indicates whether Peer Call Information and Peer Call Setup traps should be generated for all peers. If the value of this field is enabled, traps will be generated for all peers. If the value of this field is disabled, traps will be generated only for peers having Trap Enable set to enabled. Peer Configuration This table shows the list of peers from which the managed device will accept calls or to which it will place them. Double-clicking an entry in this table opens a Peer Configuration view which lets you configure that entry. This table contains the following information. Peer ID This field identifies a single peer. There may be several entries in this table for one peer, defining different ways of reaching this peer. Thus, there may be several entries in this table with the same value of Peer ID. Multiple entries for one peer may be used to support multilink as well as backup lines. A single peer will be identified by a unique value of this field. Several entries for one peer MUST have the same value of Peer ID, but different entries and, thus, different values of IF Index. Device Management Page 25

26 IF Index This entry is effectively permanent, and contains information to identify the peer, how to connect to the peer, how to identify the peer and its permissions. The value of Originate Address must be specified before a new row in this table can become active(1). Any writable parameters in an existing entry can be modified while the entry is active. The modification will take effect when the peer in question will be called the next time. An entry in this table can only be created if the associated interface entry already exists. IF Type The interface type to be used for calling this peer. In case of ISDN, the value of ISDN (63) is to be used. Lower IF Index value of an interface the peer will have to be called on. For example, on an ISDN interface, this can be the IF Index value of a D channel or the IF Index value of a B channel, whatever is appropriate for a given peer. As an example, for Basic Rate leased lines it will be necessary to specify a B channel IF Index, while for semipermanent connections the D channel IF Index has to be specified. If the interface can be dynamically assigned, this field has a value of zero. Originate Address Call Address at which the peer will be called. This can be known as the set of characters following ATDT or the phone number included in a D channel call request. The structure of this information will be switch type specific. If there is no address information required for reaching the peer, i.e., for leased lines, this field will be a zero length string. Answer Address Calling Party Number information element, for example, passed in an ISDN SETUP message by a PBX or switch, for incoming calls. This address can be used to identify the peer. If this address is either unknown or identical to Originate Address, this field will be a zero length string. Sub-Address Subaddress at which the peer will be called. If the subaddress in undefined for the given media or unused, this is a zero length string. Closed User Group Closed User Group at which the peer will be called. If the Closed User Group is undefined for the given media or unused, this is a zero length string. Device Management Page 26

27 Speed The desired information transfer speed in bits/second when calling this peer. The detailed media specific information, e.g. information type and information transfer rate for ISDN circuits, has to be extracted from this field. If the transfer speed to be used is unknown or the default speed for this type of interface is unknown, the value of this field may be zero. Info Type The Information Transfer Capability to be used when calling this peer. Choices are: speech - non-data connection audio31 and audio7 - data mode connections Permission This field is the applicable permissions. Callback either rejects the call and then calls back, or uses the Reverse charging information element if it is available. Note that callback is supposed to control charging, not security, and applies to callback prior to accepting a call. Callback for security reasons can be handled using PPP callback. Inactivity Timer The connection will be automatically disconnected if no longer carrying useful data for a time period, in seconds, specified in this field. Useful data in this context refers to forwarding packets, including routing information; it excludes the encapsulator maintenance frames. A value of zero means the connection will not be automatically taken down due to inactivity, which implies that it is a dedicated circuit. Minimum Duration Minimum duration of a call in seconds, starting from the time the call is connected until the call is disconnected. This is to accomplish the fact that in most countries charging applies to units of time, which should be matched as closely as possible. Maximum Duration Maximum call duration in seconds. Zero means unlimited. Carrier Delay The call timeout time in seconds. The default value of zero means that the call timeout as specified for the media in question will apply. Device Management Page 27

28 Call Retries The number of calls to a non-responding address that may be made. A retry count of zero means there is no bound. The intent is to bound the number of successive calls to an address which is inaccessible, or which refuses those calls. Some countries regulate the number of call retries to a given peer that can be made. Retry Delay The time in seconds between call retries if a peer cannot be reached. A value of zero means that call retries may be done without any delay. Failure Delay The time in seconds after which call attempts are to be placed again after a peer has been noticed to be unreachable, i.e., after Call Retries unsuccessful call attempts. A value of zero means that a peer will not be called again after Call Retries unsuccessful call attempts. Trap Enable This field indicates whether Call Information and Call Setup traps should be generated for this peer. Config Status The status of one row in this table. Peer Status This table shows statistics information for each peer entry. There will be one entry in this table for each entry in the Dial Control Peer Configuration Table. Peer ID Statistics information for a single Peer. This entry is effectively permanent, and contains information describing the last call attempt as well as supplying statistical information. IF Index This shows statistics information for a single peer. This entry is effectively permanent, and contains information describing the last call attempt as well as supplying statistical information. Connect Time Accumulated connect time to the peer since system startup. This is the total connect time, i.e., the connect time for outgoing calls plus the time for incoming calls. Charged Units The total number of charging units applying to this peer since system startup. Only the charging units applying to the local interface, i.e., for originated calls or for calls with Reverse charging being active, will be counted here. Device Management Page 28

29 Successful Calls Number of completed calls to this peer. Failed Calls Number of failed call attempts to this peer since system startup. Accepted Calls Number of calls from this peer accepted since system startup. Refused Calls Number of calls from this peer refused since system startup. Last Disconnect Cause The encoded network cause value associated with the last call. This field will be updated whenever a call is started or cleared. The value of this field will depend on the interface type as well as on the protocol and protocol version being used on this interface. Last Setup Time The value of System Up Time when the last call to this peer was started. For ISDN media, this will be the time when the setup message was received from or sent to the network. This field will be updated whenever a call is started or cleared. RFC2128 Dial Control Active Calls Monitor View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select RFC2128 Dial Control > Call Active. In addition to the Active Calls Table (Page 30), this view provides the following information. VoIP Current Calls The number of connections at any one time. VoIP Total Calls The summation of all calls received. Last Disconnect Text ASCII text describing the reason for the last call termination. This field exists because it would be impossible for a management station to store all possible cause values for all types of interfaces. It should be used only if a management station is unable to decode the value of Last Disconnect Cause. This field will be updated whenever a call is started or cleared. Device Management Page 29

30 Polling Interval Displays the number of seconds this table waits before the table reads and calculates the number of calls. Note: Reset VoIP Call Counter It is recommended that the polling interval be set to 20 seconds. A polling interval of 0 will turn off the current and total call counters as well as the graphing functions. This button resets the VoIP Call Counter back to zero. message was received from or sent to the network. Index ID A small index variable to distinguish calls that start in the same hundredth of a second. Peer Address The number this call is connected to. If the subaddress is undefined or not available, this will be a zero length string. Peer Sub Address The subaddress this call is connected to. If the subaddress is undefined or not available, this will be a zero length string. Active Calls Table This table contains the following information about active calls to a specific destination. Setup Time The value of System Up Time when the call associated to this entry was started. This will be useful for an NMS to retrieve all calls after a specific time. Also, this field can be useful in finding large delays between the time the call was started and the time the call was connected. For ISDN media, this will be the time when the setup Device Management Page 30

31 Peer ID This is the ID value of the peer table entry to which this call was made. If a peer table entry for this call does not exist or is unknown, the value of this field will be zero. Peer IF Index This is the interface index value of the logical interface through which this call was made. For ISDN media, this would be the interface index of the B channel which was used for this call. If the interface index value is unknown, the value of this field will be zero. Logical IF Address This is the interface index value of the logical interface through which this call was made. For ISDN media, this would be the interface index of the B channel which was used for this call. If the interface index value is unknown, the value of this field will be zero. Connect Time The value of System Up Time when the call was connected. If the call is not connected, this field will have a value of zero. Call State This is the current call state. For a list of call states and their descriptions, see Table 8. State unknown connecting connected active Table 8: Call Origin This is the origin of the call. Call States Description The call state is unknown. A connection attempt (outgoing call) is being made. An incoming call is in the process of validation. The call is active. Charged Units The number of charged units for this connection. For incoming calls or if charging information is not supplied by the switch, the value of this field will be zero. Info Type The information type for this call. Transmit Packets The number of packets which were transmitted for this call. Device Management Page 31

32 Transmit Bytes The number of bytes which were transmitted for this call. Receive Packets The number of packets which were received for this call. Receive Bytes The number of bytes which were received for this call. RFC2128 Dial Control Call History Table View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select RFC2128 Dial Control >Call History. This view provides Max Table Length and Retain Timer fields in addition to the Call History Table (Page 32), all described below. Max Table Length The upper limit on the number of entries that the Call History Table may contain. A value of 0 will prevent any history from being retained. When this table is full, the oldest entry will be deleted and the new one will be created. Retain Timer The minimum amount of time that a Call History entry will be maintained before being deleted. A value of 0 will prevent any history from being retained in the Call History Table, but will neither prevent Call Completion traps being generated nor affect other tables. Call History Table Note: This table has been replaced by the Cisco Dial Control Call History Table (Page 34). This table contains the following information about specific calls to a specific destination. Peer Address The number this call was connected to. If the number is not available, then it will have a length of zero. Peer Sub-Address The subaddress this call was connected to. If the subaddress is undefined or not available, this will be a zero length string. Peer Id This is the ID value of the peer table entry to which this call was made. If a peer table entry for Device Management Page 32

33 this call does not exist, the value of this field will be zero. Peer IF Index This is the interface index of the peer table entry to which this call was made. If a peer table entry for this call does not exist, the value of this field will be zero. Logical IF Index This is the interface index value of the logical interface through which this call was made. For ISDN media, this would be the interface index of the B channel which was used for this call. Disconnect Cause The encoded network cause value associated with this call. The value of this field will depend on the interface type as well as on the protocol and protocol version being used on this interface. Disconnect Text ASCII text describing the reason for call termination. This field exists because it would be impossible for a management station to store all possible cause values for all types of interfaces. It should be used only if a management station is unable to decode the value of Last Disconnect Cause. Connect Time The value of System Up Time when the call was connected. Disconnect Time The value of System Up Time when the call was disconnected. Call Origin The origin of the call. Charged Units The number of charged units for this connection. For incoming calls or if charging information is not supplied by the switch, the value of this field will be zero. Info Type The information type for this call. Transmit Packets The number of packets which were transmitted while this call was active. Transmit Bytes The number of bytes which were transmitted while this call was active. Receive Packets The number of packets which were received while this call was active. Device Management Page 33

34 Receive Bytes The number of bytes which were received while this call was active. Cisco Dial Control Call History Table View Access: From the Icon Subviews menu for the CiscoVoiceApp Application icon, select Cisco Dial Control Call History. This view provides the Cisco Dial Control Call History Table, described below. Cisco Dial Control Call History Table Note: This table replaces the RFC2128 Dial Control Call History Table (Page 32). This table contains the following information. Call Setup Time The value of System Up Time when the call was started. This will be useful for an NMS to sort the call history entry with call setup time. Also, this field can be useful in finding large delays between the time the call was started and the time the call was connected. For ISDN media, this will be the time when the setup message was received from or sent to the network. Peer Address The number this call was connected to. If the number is not available, then it will have a length of zero. Peer Sub-Address The subaddress this call was connected to. If the subaddress is undefined or not available, this will be a zero length string. Peer Id This is the Id value of the peer table entry to which this call was made. If a peer table entry for this call does not exist, the value of this field will be zero. Peer IF This is the IfIndex value of the peer table entry to which this call was made. If a peer table entry for this call does not exist, the value of this field will be zero. Logical IF This is the ifindex value of the logical interface through which this call was made. For ISDN media, this would be the ifindex of the B channel which was used for this call. Device Management Page 34

35 Disconnect Cause The encoded network cause value associated with this call. The value of this field will depend on the interface type as well as on the protocol and protocol version being used on this interface. Disconnect Text ASCII text describing the reason for call termination. This field exists because it would be impossible for a management station to store all possible cause values for all types of interfaces. It should be used only if a management station is unable to decode the value Dial Control Peer Stats Last Disconnect Cause. Connect Time The value of System Up Time when the call was connected Disconnect Time The value of System Up Time when the call was disconnected. Call Origin The origin of the call. Charged Units The number of charged units for this connection. For incoming calls or if charging information is not supplied by the switch, the value of this field will be zero. Info Type The information type for this call. Transmit Packets The number of packets which were transmitted while this call was active. Transmit Bytes The number of bytes which were transmitted while this call was active. Receive Packets The number of packets which were received while this call was active. Receive Bytes The number of bytes which were received while this call was active. Device Management Page 35