Kofax Communication Server Fax over IP Technical Manual€¦ · Kofax Communication Server Fax over...

Kofax Communication ServerFax over IP Technical ManualVersion: 10.3.0

Date: 2019-12-13

© 2019 Kofax. All rights reserved.

Kofax is a trademark of Kofax, Inc., registered in the U.S. and/or other countries. All other trademarksare the property of their respective owners. No part of this publication may be reproduced, stored, ortransmitted in any form without the prior written permission of Kofax.

Table of ContentsPreface...........................................................................................................................................................7Related Documentation................................................................................................................................. 8Chapter 1: Introduction.............................................................................................................................. 9

Voice and Fax over IP....................................................................................................................... 9Kofax Solution.....................................................................................................................................9Use Cases........................................................................................................................................ 10

Installation with Kofax Communication Server......................................................................10Standalone Configuration...................................................................................................... 11

International Standards and Recommendations.............................................................................. 11IP Infrastructure Requirements..............................................................................................11Compatibility...........................................................................................................................12Unicode Support.................................................................................................................... 12

Chapter 2: Installation.............................................................................................................................. 13Prerequisites..................................................................................................................................... 13Setup.................................................................................................................................................15

Personal Firewall Settings..................................................................................................... 18Standalone Installer (FoIP.msi)........................................................................................................ 19

Uninstalling KCS FoIP........................................................................................................... 20Reboot Warning When Upgrading FoIP on Windows Server 2008...................................... 20

Upgrade from Older Releases......................................................................................................... 21Upgrade from KCS FoIP 3.11.xx – 3.18.xx (KCS 9.0 – 9.2).................................................21Upgrade from KCS FoIP 3.06.xx (KCS 8.2)......................................................................... 21

Basic Configuration...........................................................................................................................21General...................................................................................................................................22List of Call Peers...................................................................................................................22

Advanced Configuration................................................................................................................... 23VoIP/FoIP Network.................................................................................................................23Voice.......................................................................................................................................26Fax......................................................................................................................................... 27H.323 Signaling......................................................................................................................28SIP Signaling......................................................................................................................... 29SSL Certificate.......................................................................................................................31HTTP...................................................................................................................................... 32KCS Integration......................................................................................................................33

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Kofax Communication Server Fax over IP Technical Manual

Storage Settings.................................................................................................................... 34Advanced............................................................................................................................... 34

Saving Configuration........................................................................................................................ 36Running FoIP in a Console Mode................................................................................................... 37Policy-based Quality of Service....................................................................................................... 38

Chapter 3: Web Portal.............................................................................................................................. 44Call Peer Statistics........................................................................................................................... 45Send Fax via Storage...................................................................................................................... 46

Sending in Loop-Back Mode................................................................................................. 47Fax Header Line....................................................................................................................48

Fax Storage...................................................................................................................................... 49Maintaining Fax Storage........................................................................................................49Inbound Messages.................................................................................................................50Outbound Messages..............................................................................................................51

Chapter 4: Integration with KCS............................................................................................................. 53Installation of Fax for KCS...............................................................................................................53

Step 1: Set Up TCOSS Server for FoIP............................................................................... 53Step 2: Set Up FoIP..............................................................................................................54Step 3: Verify Installation.......................................................................................................54Step 4: Set Up Call Peers.................................................................................................... 55

Installation of Message Wait for KCS.............................................................................................. 55Step 1: Configure KCS FoIP.................................................................................................56Step 2: Configure CallManager and Test Message Waiting................................................. 56Step 3: Create Services and Events in TCfW...................................................................... 56Message Wait via SIP Notify (RFC 3842).............................................................................57

Installation of Voice for KCS............................................................................................................ 57Install Voice Server................................................................................................................57Configure Voice Support in FoIP...........................................................................................59Enable Voice in the UFI Channels........................................................................................59Test the First Incoming Voice Call........................................................................................ 60Leave a Messages via Telephone.........................................................................................60Test Outgoing Voice Call with TC/Player.............................................................................. 60Unique Single Number Support............................................................................................ 61Hints....................................................................................................................................... 62

Supplementary Services...................................................................................................................62Call Diversion (Redirecting Number).....................................................................................62DTMF Reception....................................................................................................................64DTMF Generator for Two-Stage Outbound Dialing...............................................................64

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Call Transfer.......................................................................................................................... 65Configuration Hints........................................................................................................................... 65

Setting Own Calling Party Number....................................................................................... 65Fax on Demand.....................................................................................................................66High Availability Support........................................................................................................66SIP Registrar and H.323 Gatekeeper................................................................................... 73Behavior If an Incoming Call Is Rejected..............................................................................74FoIP and Internet Telephony Service Providers....................................................................74V.34 Fax.................................................................................................................................75Support Variable Gateway IP for Outgoing Calls..................................................................76Support of Static NAT............................................................................................................77Endpoint Name...................................................................................................................... 78T.38 vs. G.711 Pass Through Fax........................................................................................ 78T.38 Redundancy...................................................................................................................79Encryption of VoIP/FoIP Network Traffic...............................................................................80Voice Prompts........................................................................................................................87Multiple FoIP Instances......................................................................................................... 88

Configuration Values.........................................................................................................................88UFI Channel Configuration.................................................................................................... 88Description of UFI Specific Configuration Lines....................................................................92Tracing Possibilities in FoIP.................................................................................................. 93Log Entry Filtering................................................................................................................. 95

Chapter 5: Migration to KCS FoIP...........................................................................................................97Migration from KCS Line Server Model 305....................................................................................97Migration of FoIPv2 (XCAPI)............................................................................................................98

Migrate TCOSS 7.80.04 with XCAPI 3.03.36 (H.323) to FoIPv3.......................................... 98Roll Back from FoIPv3 to XCAPI.......................................................................................... 98Feature Comparison of FoIPv2 and FoIPv3......................................................................... 99

Migration of Message Wait via H.323..............................................................................................99Step 1 – Configure KCS FoIP...............................................................................................99Step 2 – Verify New Message Waiting Indication Functionality.......................................... 100Step 3 – Change the Prefix of Message Waiting Services................................................. 100Step 4 – Update Arr99 File.................................................................................................101Step 5 – Disable Old Message Waiting.............................................................................. 101

Migration of H.323 Voice Engine................................................................................................... 101Step 1 – Reinstall KCS Server Package............................................................................ 102Step 2 – Update Registry....................................................................................................103Step 3 – Configure KCS FoIP.............................................................................................104

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Hints..................................................................................................................................... 104Feature Comparison............................................................................................................ 104

Chapter 6: Administration...................................................................................................................... 105Errors Codes in TCfW....................................................................................................................105

Two-Character TCOSS Error Codes................................................................................... 105Additional Error Details........................................................................................................107

Error Codes in FoIP....................................................................................................................... 107Error Categories...................................................................................................................107Error Codes..........................................................................................................................109Connection Level................................................................................................................. 117

Event Log Entries...........................................................................................................................118Chapter 7: Development Tools.............................................................................................................. 119

Send Fax in Real Time.................................................................................................................. 119T.38 Player......................................................................................................................................120

Prepare the T.38 Fax Stream to Replay............................................................................. 121Initiating a Call.....................................................................................................................122Showing Current State of the Call...................................................................................... 123Interrupting a Call................................................................................................................ 124Fax Replay Limitations........................................................................................................ 125

T.38 PCAP Converter..................................................................................................................... 125Test Server......................................................................................................................................126

Chapter 8: Web Service Functions....................................................................................................... 127Chapter 9: Specifications....................................................................................................................... 128

Supported TIFF Options.................................................................................................................128How to Create a TIFF File for Faxing............................................................................................128

Microsoft Office Document Image Writer............................................................................ 129KCS Printer Driver...............................................................................................................129TCIMG32..............................................................................................................................130

Chapter 10: Acknowledgments..............................................................................................................131

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Preface

This document describes Kofax Communication Server Fax over IP (FOIP) solution (Using IP Call Control:V3.03). It is mainly intended for system administrators responsible for installing and configuring FoIP andVoIP in their environment.

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Related Documentation

This document refers to the following documentation:• Environment Guide - Platform System Manual• Line Server Model 305• PBX documentation (PBX Integration Technical Manual and PBX Requirements Technical Manual)• TCOSS Application Module Manual• TCOSS Configuration Manual• TCOSS System Manual• TC/FoIP Tracing Technical Manual• Voice Server documentation (Voice Platform Technical Manual, Voice xSP Integration Technical

Manual)• VoIP / FoIP Environment Guide

The full documentation set for Kofax Communication Server is available at the following location:

https://docshield.kofax.com/Portal/Products/en_US/KCS/10.3.0-SihMvq5oti/KCS.htm

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https://docshield.kofax.com/Portal/Products/en_US/KCS/10.3.0-SihMvq5oti/KCS.htm

Chapter 1

Introduction

This chapter describes fax and voice over IP solutions in general, the architecture and use cases of KofaxCommunication Server Fax over IP and international standards and recommendation.

Voice and Fax over IPSince 1990s, Voice over IP (VoIP) solutions have gradually been replacing traditional telephony. VoIPnetworks share the infrastructure that already exists for data communication. If a company has multiplelocations, data connections between these locations can be used for internal telephone calls. Establishingand maintaining a WAN infrastructure between the different locations may produce lower costs thanpaying a telephony network provider for every call.

In 1998 ITU released the T.38 recommendation dealing with the transmission of faxes over IP networksin real time. This standard guarantees interoperability between fax over IP terminals. T.38-compatiblegateways are used to adopt the T.38 protocol to PSTN (typically via ISDN) so that communication withconventional Group 3 fax machines in the public PSTN is possible. Today, T.38 is an industry standardwhich is supported even by low cost VoIP gateways.

Fax transmission is also possible over VoIP networks using the so called G.711 pass through mode. Thiskind of operation does not require T.38 support by the IP environment but it is much more sensitive fornetwork jitter, packet loss and clock synchronization problems.

Kofax SolutionThe fax over IP solution communicate with the H.323 or SIP network. It supports FoIP using T.38recommendation, FoIP G.711 pass through mode and VoIP.

Kofax Communication Server Fax over IP is composed of the following components:• FoIP core• Web interface: Allows to monitor the status and send/receive test faxes• Call peers: Responsible for communicating with T.38 compatible devices• TCOSS connection: Responsible for connection with TCOSS; distinguishes between voice and fax• Voice connection: Responsible for connection to the Voice Server• Fax storage: Acts as an inbox and outbox for faxes when TCOSS is not used

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The switch in the FoIP Core (see Basic Configuration) allows to set up the destination of incoming faxes.Incoming faxes can either be received into TCOSS or into internal fax storage of FoIP (see also the nextsection on Use Cases). Incoming voice calls are supported via TCOSS connection only.

Use CasesKofax Communication Server Fax over IP was designed for two main scenarios:• Installation with Kofax Communication Server• Standalone configuration

Installation with Kofax Communication ServerFoIP can be configured to receive incoming faxes directly into Communication Server (TCOSS). In thisscenario, FoIP can be used as a full-featured replacement of existing fax over IP solutions for TCOSS.

The scenario with TCOSS involves at least three of the five FoIP components: TCOSS connection, FoIPcore, and connection to call peers, such as a FoIP gateway. Outgoing messages from TCOSS are sentin real time (without storage) to their intended recipient. Similarly, incoming messages are delivered toTCOSS.

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Because the messages are stored in TCOSS, the FoIP internal fax storage is not used in this scenario.The web interface is an optional component as well; it can be used to monitor the status of the solutionand for troubleshooting.

FoIP can also be configured to transfer voice messages to the Kofax Voice Server. In this scenario, anadditional component is required, the Voice connection.

Standalone ConfigurationOwing to its modular architecture, for fax communication, FoIPv3 can also be used without TCOSS.Incoming faxes are stored in its internal fax storage. Outgoing faxes are queued in the fax storage andsent out. Faxes can be managed by the web interface monitoring tool.

Additionally, FoIP can provide fax functionality to other applications, such as Kofax Capture or Avalinproducts. These applications can use Web Services to send and receive faxes via FoIP. This configurationis recommendable for testing, troubleshooting or demonstrations, however, it is not supported forproductive use.

International Standards and RecommendationsFax over IP is compliant to the following standards and recommendations:• T.38: ITU-T recommendation on real time fax transmission over IP networks (Versions 0, 1, 2 and 3)• T.30: ITU-T recommendation on faxes• SIP: Session Initiation Protocol defined in RFC 3261 by IETF (UDP, TCP and TLS transports are

supported)• SIP Call Transfer: The Session Initiation Protocol Refer Method defined in RFC 3515 by IETF• H.323: ITU-T recommendation defining protocols for audiovisual communication in packet networks• DTMF via H.245 (alphanumeric and signal) and RFC 2833 (RTP-NTE)• Various call diversion standards; see Call Diversion (Redirecting Number).

IP Infrastructure RequirementsThis section defines the generic requirements to an VoIP/FoIP environment in order to support basicfunctionality with KCS FoIP. More detailed information about compatibility, configuration and hints withvarious different systems can be found in the VoIP / FoIP Environment Guide.• The IP system must support either H.323 or SIP signaling.• When using H.323, one of the following connection options must be supported:

• Integration as H.323 trunk like a standard gateway• Integration via H.323 gatekeeper as terminal• Integration via H.323 gatekeeper as gateway

• When using SIP, one of the following connection options must be supported:• Integration as SIP gateways with optional UserId/Password authentication• Integration via SIP registrar with optional UserId/Password authentication

• The IP system must support VoIP using G.711 (A-law or u-law)

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The following additional requirements are needed for fax support via T.38 only.• T.38 Version 0 (according to T.38 1998 ASN.1). According to the standard, version 0 must be supported

by all T.38 compliant devices.• The mode changes from Voice to T.38 must be signaled using H.323/SIP message (protocol based

switch-over). Proprietary methods like NSE-based switch-over (may be used by gateways connectedvia SCCP protocol to an IP environment) are not supported.

CompatibilityFoIP is compatible with the following third-party products.

VoIP/FoIP Gateways• Cisco 2600 and 2800 series*• Innovaphone• Audiocodes• Patton• Vegastream• Mediatrix

Software PABX• Cisco Call Manager 4*• Cisco Unified Communication Manager 6*• Cisco Unified Communication Manager 9• Siemens HiPath 4000 V4 (H.323 and SIP)• Siemens OpenScape Voice (former known as HiPath 8000)• Asterisk

Internet Telephone Service Provider• Sipcall (www.sipcall.com)

*For configuration, refer to the VoIP / FoIP Environment Guide.

Unicode SupportUnicode is supported for fax over IP. It is not supported for voice (for example, text-to-speech engines).

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Chapter 2

Installation

This chapter describes the installation procedure as well as general configuration.

PrerequisitesThis section provides the prerequisites for a standard installation. The actual requirements may bereduced (for example, if optional components are disabled or the size of storage/trace file is reduced).• Operating System: Any currently supported Operating System. For more details about supported

operating systems, refer to the Environment Guide - Platform System Manual.• Minimum free disk space: 100 MByte• Minimum available memory: 50 MByte + 4 MByte for each used TCOSS channel• Available network bandwidth between FoIP and gateway per channel:

• Minimum 100 kb/s during call establishment and in voice mode (before switching to T.38 mode)• Minimum 30 kb/s during transmission of T.38 data (V.17 w/o T.38 redundancy). See T.38

Redundancy for details• Network turnaround trip time between FoIP and gateway:

• Maximum 2 seconds turnaround trip time (Fax)• Maximum 400ms turnaround trip time (Voice)

• Network jitter between FoIP and gateway:• Maximum 400ms jitter for T.38 fax• Maximum 30ms jitter for G.711 pass through fax• Maximum. 60ms jitter for voice mode

Additional prerequisites if FoIP is used with Kofax Communication Server (TCOSS)• TCOSS 7.80.04 or higher• Network bandwidth between FoIP and TCOSS 7:

• Minimum 30 kb/s available bandwidth per channel• The average packet turnaround time with 260 byte block size must be:

• less than 400ms for up to 4 channels• less than 200ms for up to 30 channels• less than 80ms for up to 60 channels• less than 20ms for up to 180 channels• Single interruptions are tolerated up to 2000 ms. Longer interruptions may lead to a break in the

current transmission. If the network is interrupted for more than 9 seconds, the KCS system may

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restart the KCS FoIP. Note that this 9s time-out may be increased as described in the TCOSSConfiguration manual chapter “Network interruptions to LS1 and FoIP”.

• The numbers above describes the network requirements for channels with up the 14.400 bps (V.17). AV.34 (were up to 33.600 bps are used) channels has the same requirement as two V.17 channels.

• Up to 180 channels (V.17 or V.34) are supported on a single physical computer when using T.38 Fax.Up to 120 channels (V.17 or V.34) are supported on a single virtual machine when using T.38 Fax. Upto 120 G.711 passthrough mode channels are supported on physical or virtual machines.

• The following CPU resources must be available.

FoIP usage [SPECInt_Rate_Base2006] per channelChannel type

Physical machine Virtual machine

T.34 fax channels with V.17 mode 1.0 1.0

T.34 fax channels with V.34 mode 1.5 1.5

G.711 pass through mode 2.0 2.5

Voice channel 1.5 1.5

1. CPU performance results for many systems are published at www.spec.org. Start search at http://www.spec.org/cgi-bin/osgresults?conf=cpu2006 and use the SPECInt_rate_base2006 value froma system that is similar to your hardware.

2. The maximum number of channels can be calculated with the following formula: (CpuPerformance - ReservedForOtherProcesses) MaxChannels=---------------------------------------------- FoipUsagePerChannel

Here are some typical example values.

Maximum number of channels1Hardware Deployment2

T.38 (V.17) G.711

HP ProLiant ML350 G5 (1x X5355 Quadcore, 2.66 GHz, SCPECInt_rate_base2006= 44)

Physical machine 30 15

HP ProLiant DL380G5 (2*X5420 Quadcore, 2 GHz, SCPECInt_rate_base2006 =80)


Virtual machine (1/12 cores) 30 10





HP Proliant DL380G8 (2*Xeon E5-2640 6-core, 2.5GHz, SCPECInt_rate_base2006 =445)


1. The maximum number are channels are rounded under the assumption that up to 20% from totalavailable CPU may be consumed by other high priority processes on the same virtual/physicalmachine.

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http://www.spec.org/cgi-bin/osgresults?conf=cpu2006

http://www.spec.org/cgi-bin/osgresults?conf=cpu2006


2. On virtual machines this column also indicates the number of assigned CPU cores.

3. Maximum number of channels is additionally restricted by 120/180 channel limit as described above.

• The operation with G.711 pass-through mode is real-time critical. In case of insufficient CPUperformance you may receive send/reception errors with TCOSS error code JO, error categoryBAD_LOCAL_PERF (LPERF).

• The support of G.711 pass-through channels on virtual machines critical due to real-time requirements.VoIP gateways generate a signal dropout on jitter buffer overrun/underrun. Such a dropout mayinterrupt the fax call (RTP packet generation must not be deferred more than about 50ms). At least2 cores should be reserved for optimum quality. Refer to the Environment Guide - Platform SystemManual for further details.

Additional prerequisites if FoIP is used for voice with Kofax Communication Server (TCECP)• TC/SP 7.86.00 or higher• Network bandwidth between FoIP and TCECP server per channel:

• Minimum 150 kbps available bandwidth• Maximum 0.5 seconds turnaround trip time

• Up to 30 concurrent voice calls are supported with a single FoIPv3 application.

SetupKCS FoIP is installed as a part of KCS server package.

If you had previously installed the standalone variant of KCS FoIP (using FoIP.msi), you have to uninstallthis application. See Standalone Installer (FoIP.msi) for more information about the standalone installer.

If you decide to preserve existing configuration when uninstalling the standalone variant, you can laterimport this configuration into FoIP included in server package.

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1. Start the setup by launching SETUP.exe. From the Fax over IP (Group), select FoIP.

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If you have previously used the standalone variant of FoIP on the same computer, and if you opted topreserve its configuration, you can now import it.

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2. Enter TCSRV settings and click OK.

KCS FoIP is installed into the C:\TOPCALL\FOIP\00\ directory.After the installation, KCS FoIP configuration opens. See Basic Configuration and AdvancedConfiguration.The following new shortcut is added into the start menu:

Personal Firewall SettingsIf you are using a personal firewall on your machine, you may get a warning as shown in the followingfigure.

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Either disable the personal firewall or click Unblock to allow the Merlin Process to access network.

Note Merlin Process is the framework used by FoIP.

Standalone Installer (FoIP.msi)Before KCS 9.0, KCS FoIP was delivered as a standalone installer, FoIP.msi. This section lists thedifferences to standard installation.

The default installation directory is C:\Program Files\Kofax\Fax Connector\FoIP\.

After a successful installation, the following new shortcuts are added into the start menu.

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Start Menu shortcuts

Shortcut Target of the shortcut Description

KCS FoIP ConfigurationUtility

C:\Program Files\Kofax\Fax Connector\FoIP\00\Configure.bat

See Basic Configuration.

KCS FoIP Status Monitor http://localhost:25081 See Web Portal.

KCS FoIP Trace Directory C:\Program Files\Kofax\Fax Connector\FoIP\00\trace

Useful for troubleshooting.

Uninstalling KCS FoIP1. Do one of the following to uninstall the application:

• Start the installation package again and click Remove.• In the Control Panel, click Add or Remove Programs, select Kofax Communication Server Fax

over IP and click Remove.

The following window is displayed:

2. Click No to delete all files created by KCS FoIP or click Yes to keep configuration files and the faxstorage file on the disk.

Reboot Warning When Upgrading FoIP on Windows Server 2008When FoIP is upgraded by a normal user (not the built-in “Administrator” account) on Windows Server2008 where UAC (User Account Control) is active, the following reboot warning is displayed during theinstallation:

You can ignore the displayed warning. No reboot is necessary. Click OK.

You can also stop the KCS FoIP service before starting the upgrade. The reboot warning will not appear.

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Upgrade from Older ReleasesAs a general rule, an upgrade works like a new installation with the difference that the existingconfiguration is used. This chapter describes some hints or additional manual steps during upgrade.

Upgrade from KCS FoIP 3.11.xx – 3.18.xx (KCS 9.0 – 9.2)• The QoS Configuration for SIP/H.323 signaling (which was supported up Windows 2003/XP only) is

no longer supported. It is expected that all installations are working fine with the default DSCP for SIP/H.323. If there is any demand to set a DSCP value for SIP/H.323, the policy-base option as describedin chapter 3.9 can be used on Windows 2008/Vista or later.

Upgrade from KCS FoIP 3.06.xx (KCS 8.2)• All hints described for the Upgrade from KCS FoIP 3.11.xx above.• All trace files are deleted during update.• If you are using the Voice integration with KCS and a PABX prefix is configured, position 2 of

configuration line 295 in all UIF channels must be set to 01 in order to use the configured prefix withoutgoing voice call.

• If you are using the Voice integration with KCS via SIP, the reception of via RTP named-telephone-events according to RFC 2833 must be enabled in the FoIP configuration. Otherwise DTMF input willnot work after update.

• If you migrate from standalone installer (FoIP.msi) to KCS FoIP included in server package, see Setup.

Basic ConfigurationAfter the initial installation, you can configure the application. If FoIP had been installed on the computerbefore (and the installation directory has not been cleared or changed), the previous configuration is usedautomatically.

Select the “Configure FoIP (FOIP)” shortcut from the Start menu.

A browser window with the following configuration form opens.

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The first two tabs, General and List of Call Peers of the configuration are expanded by default. Thesetabs contain the most important settings that must be configured before use.

The additional tabs contain advanced settings and the default values usually do not need to be modified.

GeneralIn this section, select the destination for inbound documents.

Value Description

Description You can use any text to describe this instance. This text is shown in the Web-status page andKCS Monitor. It may be used for a better overview if multiple FoIP instances are used. It has nofunctional impact on the operation of KCS FoIP.

Reception Mode Defines the target of inbound calls. You can choose either option:• TCOSS: Use this option when using FoIP with Kofax Communication Server, Fax / Voice

reception is configured in the number conversion table of the respective UFI channelconfigurations in TCOSS.

• Fax Storage: Use FoIP’s internal fax storage.

List of Call PeersIn this section, define fax servers, gateways, and other T.38 compatible devices (including, for exampleCisco CallManager) by configuring the following settings.

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Value Description

Enabled If selected, enables a given call peer.

Protocol Lists the possible signaling protocols (H.323, SIP, H.323 with RAS, or SIP withregistration). Select a protocol compatible with your call peer.

Note If you select an H.323 Gatekeeper (“H.323 with RAS”) you must not useH.323 with any other active call peer.

Remote Address:HostPort

The remote IP address or host name of the call peer (Gateway, Gatekeeper,Registrar or CallManager) and the relevant signaling port number.

Note This value supports host names for H.323 only.

Authorization:User IDPassword

The user name and password for authentication with an H.323 gatekeeper or a SIPserver. With H.323, the User ID is also us55ed as endpoint name. See EndpointName for details.

Reg. Numbers A number to be registered at an H.323 gatekeeper or SIP registrar.When the RegistrationType parameter is set to “Gateway”, the number is interpretedas a prefix. When the RegistrationType parameter is set to “Terminal”, FoIP registersas a terminal with this number at the gatekeeper.

arrows Use the arrows to change the order of the call peers in the table.

Advanced ConfigurationThis section describes advanced configuration.

VoIP/FoIP Network

Value Description

Local IP The local IP address for FoIP network. If left empty, uses all available networks.

MediaPortLow Lower limit of port range for voice/T.38 media data (>1023)

MediaPortHigh Upper limit of port range for voice/T.38 media data (<65336)

Enable QoS If selected, DSCP (Differentiated Services Code Point) value will be set.

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Value Description

DSCP for Fax DSCP value for (T.38 and pass through) fax packets.

DSCP for Voice DSCP value for voice packets.

The differentiated services code point (DSCP) values inserted into the IP header can be configuredseparately both for fax (T.38 and for G.711 pass through mode) and voice data. The “Diffserv/DSCPmodel” is defined in IETF RFC-2474 and RFC-2475.

To support DSCP, you must select the following Windows specific prerequisites:• For Windows 2000 or Windows Server 2003: Create/set registry REG_DWORD value HKLM\System\CurrentControlSet\Services\Tcpip\Parameters\DisableUserTOSSetting to "0".Refer to the knowledge base article Q248611 for more details.

• For Windows Vista/2008 or later: Install the Windows feature “Quality Windows Audio Video Experience(qWAVE)”. If qWAVE is not installed, a trace entry such as “SetDSCP(56) LoadLibrary qwave.dll failedLastError 126“ appears in the FoIP trace and the configured DSCP values are ignored. You can checkand install qWAVE with Server Manager features as shown in the following figure.

Typical DSCP values are described in the following table.

DSCP value Description Typical use

0 Best effort Default if QoS is not used at all

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DSCP value Description Typical use

8 Background, priority below best effort Data backup

26 Assured forwarding Class 3, Low Drop (AF31) H.323/SIP signaling

40 Class Selector 5 (CS5), compatible with IPprecedence 5, excellent effort

H.323/SIP signaling

46 Expedited forwarding (EF) Real-time data like VoIP voice or T.38media streams.

56 Class Selector 7 (CS7), compatible with IPprecedence 7

Real-time data like VoIP voice or T.38media streams.

Note• Under Windows Vista/2008, only DSCP values 0, 8, 40 and 56 can be set.• Under Windows 7, Windows Server 2008 R2 or later, any DSCP value may be set. However, for

values other than 0, 8, 40 and 56, the calling application must be a member of the Administrators orthe Network Configuration Operators group.

• Under Windows Vista/Windows 2008 or later, the “DSCP for Fax” value works only if the “Local IP”value is also set.

You can verify the QoS settings with Wireshark. Take a trace, select an outgoing media packet and thenexpand the Internet Protocol section. You should see the DSCP configured:

Alternatively DSCP values may also be set through group policies. See Policy-based Quality of Service.

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Voice

Value Description

MediaSecurity Enables SRTP (secure RTP) data for G.711 Pass through FoIP or voice mode.The following modes are available:• [1] disabled (use RTP, reject SRTP)• [2] negotiate (try SRTP, fallback to RTP)• [3] always (use SRTP, reject RTP)

Note SRTP is available with SIP calls only. H.323 calls are always usingRTP. When using SRTP the sip message must be encrypted as well (byusing SIP via TSL/TCP) in order to protect the SRTP session keys.

MediaSecurityCryptoSuites Offers Crypto suites in outgoing SDP. All supported suites are accepted whenoffered by remote side regardless of this configuration parameter• [1] offer only crypto suite AES_CM_128_HMAC_SHA1_80• [2] offer only crypto suite AES_CM_128_HMAC_SHA1_32• [3] offer crypto suites AES_CM_128_HMAC_SHA1_80 and

AES_CM_128_HMAC_SHA1_32 (default)

MediaSecurityUnencryptedSrtp Offers Crypto parameter UNENCRYPTED_SRTCP in outgoing SDP. Crypto withand without UNENCRYPTED_SRTCP is accepted when offered by remote sideregardless of this configuration parameter.• [1] offer only crypto without UNENCRYPTED_SRTCP• [2] offer only crypto with UNENCRYPTED_SRTCP• [3] offer crypto with and without UNENCRYPTED_SRTCP (default)

Silence Suppression Enables RTP silence suppression for voice mode.

Enabled If selected, enables a payload codec.

Codec Determines which codec type is used.

Max. Packet Interval Determines the maximum packet interval in ms for voice mode.

Note The FoIP G.711 pass through uses fix packet interval of 20ms and no silence suppression.

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Fax

Value Description

OutboundDtfmfMode Defines how to generated DMTF digits during outbound fax calls:• 0: Send DTMF digits inbound as G.711 audio• 1: Send DTMF digits are RFC 2833 telephone events if the remotes indicated to

support it. Otherwise use G.711 audio.

Select a DTMF mode that is compatible with your environment. The default valuegenerates DMTF signals as audio tones which should work fine for testing and calls viaPOTS. If the receiver expects DTMF as RFC 2833 (like KCS FoIP), then option 1 will berequired (unless DTMF signals are converted by the any DTMF relay within the network)

OutboundT38Mode Defines how to switch to T.38 mode during outbound calls. The following four modes areavailable:• 10: Immediately switch to T.38 and disconnect if T.38 is not available• 20: Use G.711 pass-through but try switch to T.38 after 1st CNG• 40: Use G.711 pass through unless T.38 is requested by remote side (default)• 60: Use G.711 pass through and prevent switch to T.38

Mode 10 has the best performance and requires the lowest bandwidth. It works only ifT.38 is supported by your environment. Modes 20 and 40 improve interoperability butthey may require better network conditions if G.711 pass through mode is used. Mode 40is used by default because it is the most compatible mode. Mode 60 should be used onlyin the rare case that T.38 must be prevented for any reason.

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Value Description

InboundT38Mode Defines how to switch to T.38 mode during inbound calls. The following options areavailable:• 10: Immediately switch to T.38 and disconnect if T.38 is not available• 15: Immediately switch to T.38 with a fallback to G.711 pass-through. This mode may

causes an additional delay of about 5s with H.323 calls in case of fall-back to G.711. Itis fully supported with SIP.

• 20: Use G.711 pass-through but try switch to T.38 after CED• 40: Use G.711 pass through unless T.38 is requested by remote side (default)• 60: Use G.711 pass through and prevent switch to T.38

Mode 10 has the best performance and requires the lowest bandwidth. It works only ifT.38 is supported by your environment. Modes 15, 20 and 40 improve interoperability butthey may require better network conditions if G.711 pass through mode is used. Mode 40is used by default because it is the most compatible mode. Mode 60 should be used onlyin the rare case that T.38 must be prevented for any reason. Mode 15 if fully supportedwith SIP, but the fax prompt is delayed by about 5s if the fallback to G.711 happens viaH.323.

EnableV34 Enables support for V.34 fax. This check box also enables the support of T.38 version 3,which is a requirement for V.34 operation. The T.38 version 3 uses a different encodingwhich is not correctly displayed by current releases of Wireshark Network ProtocolAnalyzer (tested with 1.0.7).

Note V.34 fax is not supported with G.711 pass through mode. See V.34 Fax forsupport FoIP inbound/outbound T.38 modes.

RedundancyLS Redundancy for transmitted low speed T.38 packets, that is, V.21 and all non-dataevents. See T.38 Redundancy for details.

RedundancyHS Redundancy for transmitted high speed T.38 packets, that is, V.27, V.29, V.17 and V.34data. See T.38 Redundancy for details.

H.323 SignalingThis section described global H.323 signaling parameters, which are valid for all H.323 call peers.

Value Description

Local H.323 Port The local H.323 signaling port.

RegistrationType Type of gatekeeper registration (terminal/gateway) where the type gateway is requiredif you want to support multiple inbound numbers.

ZoneId Optional zone ID used for gatekeeper registration as gateway.

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Value Description

NatTraversalIp If H.323 calls are forwarded via static NAT, you must set this value to the mapped ownIP address (as it appears on the other side of the static NAT router).

H245Tunneling If selected, enables tunneling of H.245 data over H.225 connection. This tunnelingavoids additional TCP/IP connections for H.245. It may be required for calls via NATrouter.

SIP SignalingThis section describes the global SIP signaling parameters, which are valid for all SIP call peers.

Value Description

SipEnabledTransports The transports that are enabled. Any combination of the following transports issupported:• UDP: None secured SIP via UDP• TCP: None secured SIP via TCP• TLS: SIP via TLS (transport layer security), best effort TLS• SIPS: Secure SIP via TLS (force TLS on all routes)

KCS FoIP accepts incoming requests from any enabled transport. If TLS or SIPSis selected, an SSL certificate must be configured in the SSL section (as it is withTWS, Message Connector, and others).The difference between SIPS and TLS is that SIPS requires encryption (TLS) on allroutes within the SIP network. A SIPS call should fail by design if any required route(such as SIP connection between software PABX and gateway) does not supportTLS. TLS provides the same level of encryption for the SIP messages betweenFoIP and it call peer, but it also allows to use unsecured sip connections behind thecall peer.FoIP up to 3.23.xx (KCS 10.0) supported TLS only without SIPS. This configurationoption is still included for unexpected compatiblitiy issues during update, but it isdepcreciated and may be removed in futures releases because it does not confirmto RFC 5630.SIPS is automatically enabled during update to FoIP 3.24.02 (or higher) if TLSwas enabled. This means that depriciated values are automatically converted tocompatible RFC 5630 compliant values.

SipOutgoingTranport The transport used to outgoing requests. The selected transport must be enabled.

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Value Description

Local UDP and TCP Port The local UDP and TCP port for unencrypted SIP signaling.

Local TSL Port The local TLS (over TCP) port for encrypted SIP signaling.

CheckCertificate If selected, SIP via TCP/TSL from remote side must have a valid SSL certificate/key according to the SSL Certificate configuration in order to prevent a man-in-the-middle attack.

CalledSipDomain SIP domain for called number ("to" field) with outgoing calls. If this value is notspecified, the gateway Host/IP is used.

LocalSipDomain SIP Domain for own number ("from" field) with outgoing calls. If this value is notspecified, the selected local interface IP (or configured local IP) is used.

Asserted Identity If selected, adds the originator (such as header field "from") also as P-Asserted-Identity (PAI) SIP header with outgoing calls. This may be required between trustedSIP servers to assert the identity of authenticated users.

InboundNumberSource Defines how the received inbound number is extracted from the initial SIP INVITEmessage:• Rq: Get number from the request-url. This is usually the final routed number

(default)• To: Get number from the to-header-field. This is usually the original called

number

EnableRtpNte If selected, enables reception of DTMF digits via RTP named-telephone-eventsaccording to RFC 2833. This function is required if a voice integration via SIP isused in order the allow DTMF input. It is disabled by default because it may causecompatibility problems (even with fax calls) if the IP system does not support RFC2833.

Add media for T.38 Defines if a new SDP media should be added if KCS FoIP requests a mode changefrom audio to T.38 mode:• No: Audio media is changed to T.38 (default)• Yes: Audio media is kept as inactive media and a new media stream is added for

T.38. (depreciated option)

This option exists for compatibility with option “Add inactive SDP medias” in priorreleases. It has been changed due to fix of bug 672110 so that RFC 2327 compliantpart is always included because option “Yes” may cause compatibility issues withsome none-RFC compliant implementations.

Retry RequestT38 The behavior if mode change to T.38 is rejected with SIP status 488. The followingvalues are supported:• [0] No: KCS FoIP does not retry to request T.38 mode (continue with G.711)• [1] Yes: Make one retry to request T.38 mode (default)• [2] Refresh G.711 mode: Generate a Re-Invite using G.711 mode

MulticastAddress Optional. An IPv4 multicast address. Reception via multicast address can be usedfor failover and load balancing installations. See Inbound SIP Failover with MulticastAddresses for more details.

Note IPv6 support is disabled if a multicast address is specified.

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Value Description

MulticastPeerAddresses Blank separated list of addresses (IP[:port]) that are notified after establishedmulticast inbound call. The special value 'my-group' means own multicast IP.

Note• In order to specify the originator and recipient of a call instead of plain telephone numbers, the SIP

procedure defines URLs. FoIP supports the SIP/SIPs URI schema because they are mandatory for allRFC compliant SIP implementations.

• The SIP/SIPs URI schema includes a user information and sip-domain (user-info@sip-domain) verysimilar to email addresses. The user information is usually the telephone number and the sip-domainis indented as an administrative domain that knows how to handle their numbers.

• KCS ignores the SIP domain with received requests, so that, it does not expect a specific sip-domainconfiguration in the remote call-peer. Most implementations do the same, so that, it is not required toconfigure the FoIP SIP domain. This is also the behavior of all KCS versions before KCS 10.1.

• If the used call-peer expects a specific SIP domain, it must be set in the configuration valuesCalledSipDomain and/or LocalSipDomain. Note that, it is typical but not required that the samedomain is used for both values.

SSL CertificateThe SSL Certificate parameter are used for SIP via TSL/TCP only.

Value Description

SSL Certificate Your SSL server certificate in PEM format (Base64 encoded, including -----BEGINand -----END lines)

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Value Description

EnSSL Private Key abled The private key to the above server certificate, in PEM format (Base64 encoded,including -----BEGIN and -----END lines). The private key entered must not beencrypted, it will be encrypted internally.

SSL Chain Certificate Optional intermediate certificate in the certificate chain to a well-known rootcertificate in PEM format (Base64 encoded, including -----BEGIN and -----END lines)

SSL Trusted CA Certificates Optional trusted CA certificates for verifying remote peer certificates. (Base64encoded, including -----BEGIN and -----END lines). These certificates are requiredonly if the SIP configuration value CheckCertificate is enabled.

SSL Config For SSL security configuration, select the of the following:• [10.3] TLS 1.1-1.3; OWASP-C, FS: Supports TLS 1.1, 1.2 and 1.3. Also, support

OWASP Cipher String 'C' and forward secrecy. This is default configuration.• [10.2] TLS 1.0-1.3; HIGH: Supports TLS 1.0, 1.1, and 1.2. This configuration

(including cipher list) provides the behavior of KCS 10.2.0 as good as possible.• [Adv] Advanced settings from Create_Config.xslt: This option is for advanced

users or can be used for troubleshooting. The configuration is derived from theAdvancedOpenSslConf variable in Create_Config.xslt.

HTTP

Value Description

Enabled If selected, the web portal and web interface are enabled.

Local IP Address IP address of the local interface used for web portal. If empty all local interfaces are used.

Local Port Specifies the port number of the FoIP web portal. The default value is 25081.

SSL Active If selected, activates the Secure Sockets Layer protocol (https).

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KCS Integration

Value Description

Enabled If selected, FoIP can be controlled by a TCOSS server. See Integration with KCS forfurther details.

Local IP Address IP address of local interface used for connection to TCOSS / voice server. If empty alllocal interfaces are used.

Local Port TCP listener port for connection from TCOSS.

Password Password for connection from TCOSS. If empty, the password is not checked.

CheckCallPeer A call-peer check may be activated for improved fail-over scenarios. If active thenthe connection with TCOSS is only possible if at least one of the checked call-peersis reachable. Use this feature only if an alternative FoIP Lanlink IP is configured inTCOSS.

MessageWait Method of Message Waiting Indication signaling (MWI).

Call Diversion Mode Defines the priority if multiple call diversion numbers are available.

EnabledVoiceServer If selected, the component may be controlled by a voice server.

Local Port TCP Listener port for connection from voice server.

Call Transfer Mode Select when is a call transfer considered completed: after the transfer to party hasreached Alerting or Connected state.

Call Transfer with Hold If selected, call hold is executed prior to the call transfer. This option might berequired for certain PBX.

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Storage Settings

Value Description

Enabled If selected, FoIP fax storage is available

Storage Size Size of disk space in MB reserved for storage of received faxes

Own Fax Number Called Station Identification (CSI) for incoming calls. Should contain only digits, '+'and blanks.

Bearer Capability Q.931 bearer capability for outgoing calls. This setting is only effective for H.323-calls.

Enable ECM transmission If selected, T.30 Error Correction Mode (ECM) is enabled for outgoing messages. Itis relevant for all messages posted via web-service interfaces. The ECM mode formessages transmitted via Test page can be selected directly on the test page.

Enable ECM reception If selected, T.30 Error Correction Mode (ECM) is enabled for incoming messages

Advanced

Value Description

Single Process If selected, all components of the application are running in the same process.Do not check this option if you plan to work with policy-based QoS.

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Value Description

Trace Level General trace level:• 0: Off• 10: Very rare tracing that happens only during initialization / startup• 20: Few trace data per document. Less trace output with productive systems• 30: More intense tracing. Productive use is possible but trace files can be

large• 60: Restricted performance for productive use (max. 10 channels

recommended)• 100: Extensive tracing for troubleshooting. Operating multiple channels can

fail

Message Trace Size Message size limit in bytes to trace traffic between components:• 0: Off• 1: Single line per message• More than 1: Size limit in bytes

OpenH323 Trace Level Open H323 library trace level (0..255): 0 = off, higher values give a more detailedtrace.

Note Note: OpenH323 trace does not use the trace settings (Size of TraceFile, Number of Trace Files and Append Trace). It is always appended tofile 00\trace\OpenH323.log. It’s size is not limited and may fill up yousystem of this trace is active!

BtrTraceLevel Enable generation of binary trace files for FoIP G.711 pass through mode:• 0: Off (default)• 2: Keep binary traces for failed calls only (successful calls are deleted at end

of call and therefor cause a gap in the file numbers)• 3: Keep binary traces for all calls.• 18: Like mode 2 but succesfull calls do not causes a gap in the file numbers.

Files are temporary generated with “ActiveFax” prefix and renamed to its finalname at end of call. Names are sorted by end-time. “ActiveFax” files from aprevious abnormal termination of FoiP are deleted with the first call after FoIPrestart.

BtrTraceFiles Maximum number of binary trace files. These files are created in the FoIP tracefolder with the name “fax_####.btr” where #### is a decimal number that startswith 0000 after FoIP startup and increments with each call. If the max. number oftrace files is reached the counter restarts with 0000 and overwrites old files.Note that the number of files may be exceeded by the number of active calls ifBtrTraceLevel is set to 18.

BtrTraceMaxFileSize Maximum size of binary trace files in kBytes. If this limit is exceeded furtherbinary trace output will be ignored.

Trace Location Drive and path of trace location (relative paths are based on FoIP installationfolder).

Size of Trace File Maximum size of trace files in kBytes.

Number of Trace Files Maximum number of generated trace files.

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Value Description

Append Trace If selected, appends to existing traces after restart of application.

Saving ConfigurationThe configuration options are automatically validated. Invalid / contradictory values are marked red.Erroneous configuration cannot be saved.

After modifying the configuration values, click Save to save the changes. The following screen isdisplayed:

You can either return to the configuration screen or exit the configuration. To activate the newconfiguration, use KCS Monitor to restart FOIP.

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Running FoIP in a Console ModeIt is possible to launch FoIP in a non-service mode, as a console application. Simply run the batchfile “C:\TOPCALL\FOIP\00\Run.bat” (or “C:\Program Files\Kofax\Fax Connector\FoIP\00\Run.bat” for standalone FoIP). It can only be launched when the Kofax Communication Server Faxover IP Windows service is stopped.

When using the non-service mode, the following status window is displayed after starting FoIP.

The console mode does not require any registration on the system, that is, you can copy your FoIPinstallation (such as C:\Program Files\Kofax\Fax Connector\FoIP\00) and run it at any timelater directly from that backup location (such as, USB memory stick). It uses the original configuration,

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program files and messages in the fax storage. But for security reasons, all passwords stored in theconfiguration are usable only on the same Windows where the configuration has been saved.

Policy-based Quality of ServiceNetwork administrators can use QoS mechanisms to prioritize outgoing network traffic by setting a DSCPvalue. In order to prioritize traffic in this way, you must create a series of QoS policies.

Alternatively, a DSCP value may be set in the configuration, see VoIP/FoIP Network. Policy-based DSCPsettings have priority over DSCP settings by the application (through configuration).

Note The Windows feature, “Quality Windows Audio Video Experience (qWAVE)” must be installed (asdescribed in VoIP/FoIP Network) in order to support QoS (either via FoIP configuration or policy-based).

Use Group Policy Management Console (GPMC) or Group Policy Object Editor in Windows Vista orlater to set the DSCP value based on the sending application executable name, source / destination IPaddress, port number or range, and protocol (TCP or UDP).

1. To open the local Group Policy object, click Start > Run.2. Type gpedit.msc and click OK.

3. Navigate through the console tree to Computer Configuration > Windows Settings > Policy-Based QoS.

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4. Right-click Policy-Based QoS and select Create New Policy from the menu.The Policy-Based QoS Wizard starts.

5. Enter the policy name and specify a DSCP value.Do not specify an outbound throttle rate.

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6. Click Next and specify the applications to which the policy should apply.

This table shows which executable names and protocols may be used to select a particular datastream.

Executable Name Protocol DSCP setting for

FoIP_H323.exe TCP H.323 signaling

FoIP_SIP.exe TCP and UDP SIP signaling

FoIP_T30.exe UDP T.38 and pass through fax packets

FoIP_Voice.exe UDP voice packets

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7. Click Next and enter the source and destination IP address information.By default, the policy applies to all source and destination IP addresses, but you can specifyindividual IP addresses or address ranges.

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8. Click Next and specify the protocols and ports that the policy should apply to.

The Protocol can be selected here as “TCP”, “TCP and UDP” or “UDP” (see the table on executablenames and protocols above). It is not required to enter specific port numbers or ranges.

Note• Do not select the “Single Process” option in the advanced configuration (see Advanced) if

working with policy-based QoS, the different process names are required for proper selection ofdata streams.

• We strongly recommend that you configure the QoS-policy based values only in the “ComputerConfiguration” so that it is used by all users (including system account). The “User Configuration”should not have any FoIP related entries. See the following figure.

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Chapter 3

Web Portal

This chapter describes the web monitoring tool and the functions it offers.

Once the Kofax Communication Server Fax over IP is set up, it is automatically launched as a Windowsservice.

The application provides a web portal. You can access this portal via any standard internet browser usingthe following address: http://<Computer>[:Port]/

If you are using default values, the address is: http://localhost:25081/.

On this page, you can perform the following actions:• Status: State overview of the application. This screen is displayed automatically after launching FoIP

monitor page.• TCOSS Calls: Displays the statistics of used call peers. See Call Peer Statistics.• Test: Allows to send faxes by adding them to a queue in the storage. See Send Fax via Storage.• Inbound Status; Outbound Status: Access the various folders of the storage database. See Fax

Storage.

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Call Peer StatisticsClick TCOSS Calls from the menu to display the Call Peer Statistics window. This window allows to viewthe statistics of calls on each of the configured call peer.

Each line represents a single call peer. The following table provides a more detailed description of thecolumns.

Column Description

Nr Number of the call peer as configured in FoIP configuration. Only enabled call peers are listed.

Protocol Protocol of the call peer, that is, SIP or H.323.

Last State The last known state of the call peer:

State Description

Unknown (white) The call peer is not yet used.

OK (green) The call peer was OK during the last call. This stateincludes calls where:• the remote side was busy.• the remote side did not answer the call.• fax transmission or reception failed for any reason.

Unreachable (red) Call peer unreachable, such as network down.

No Channel (yellow) In the last call attempt, the call peer was online but ithad no free channels.

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Send Fax via Storage1. Click Test from the menu to display the following screen.

The following table provides a more detailed description of the parameters.

Value Description

Number to Dial Destination fax number is the only required parameter.

Protocol / Call Peer Signaling protocol or configured call-peer that should be used.

Gateway Host Host name / IP address and optionally a port number of an H.323/SIP gateway.This parameter is not available if you are using one of the call peers from theconfiguration.

Caller ID Calling party number.

Expected CSI Expected called subscriber identification, or a part of it, for verification that thecorrect destination has been called.

Source TSI Originating subscriber identification.

Error correction mode If selected, T.30 error correction mode is used (if supported by remote side).

Max. transmissionspeed

Maximum transmission speed and supported modulations.

Note V.34 must also be enabled in the configuration (Fax tab) to use maximumspeed. See V.34 Fax for details.

Sender Name Name of the sender displayed in the fax header line.

Retry on error If selected, the fax is retransmitted if send operation fails.

TIFF File TIFF file to be sent (a built-in default file is sent if you leave this field empty). SeeSupported TIFF Options for format details.

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2. To send a test fax, proceed as follows:

a. Enter the destination fax number (required).

b. Enter optional parameters, such as the sender's name to be displayed in the fax header line.

c. Either choose a TIFF file that should be faxed or leave the field blank to use a default file.

d. To preview the message including the header line, click Preview.

e. Click Send Fax to send the message.The following message is displayed in the bottom frame: “Server returned ok!”.• The messages that are currently sent or are queued for sending are stored in the Outbound

Status – Pending folder of the fax storage.

• When a message is successfully delivered, it is moved from the Pending folder into theProcessed folder of the fax storage.

• When a message is not delivered successfully, it is moved to both Processed and Failuresfolder.

Sending in Loop-Back ModeTo test the FoIP send and receive functionality without a gateway you can do a loop-back test. This testrequires that the Reception Mode is configured to Fax Storage. See General.

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The procedure is the same as described in Send Fax via Storage, except as follows:1. Enter any number, such as 0 in the Number to Dial field.

2. Select "h323 Gateway" or "sip Gateway" from the Protocol/Call Peer list.3. Enter localhost, that is “127.0.0.1” in the Gateway Host field.

The Gateway Host value overrides the gateways set in the configuration.As 127.0.0.1 is the loop-back IP address, the call is routed back to the local FoIP application.

4. Wait until the message is delivered, then look for the received fax message in the Inbound Status –Pending folder.

5. Click the View source as XML icon on the received message and verify if there is also the loop-back IP address in the Debug information.

Fax Header LineWhen sending a fax, a header line with the most important information is added to the uppermost part ofeach page. The header line provides the following information:• To: Displays an identification of the destination fax machine. This can be either the dialed number or a

value configured on the fax machine. In Preview mode this is always the dialed number.• From: Displays the sender name.• Fax: Displays the originating subscriber identification.• Company logo.• Date and time of sending (using the local time zone of the computer where FoIP is running).• Current page and the total number of pages.

The font “Microsoft Sans Serif” used in the fax header line supports characters from several differentalphabets including Latin, Latin Extended, Greek, Cyrillic, Thai, Hebrew, and Arabic (with the restrictionthat Hebrew and Arabic is written left-to-right).

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Fax StorageIn the fax storage, all faxes which have been sent or received via storage are stored. Messages sent orreceived via TCOSS are not saved to the storage. This storage is divided into several folders, which canbe accessed by clicking on the corresponding menu item.

Up to 10 faxes can be displayed per page. Use the links in the top and bottom bar to navigate within astorage folder. Use the Refresh button to update the list.

Maintaining Fax StorageFax storage does not require maintenance. Processed faxes are automatically deleted after a certainthreshold is reached. Select Status from the menu to display information about the fax storage.

The storage status line provides the following information:• Number of pending messages. These messages are not deleted automatically. Fax storage can be

filled with pending messages; if this happens, no further messages can be received until currentmessages are processed.

• Number of processed messages. These messages are deleted as necessary to meet disk usage target.• Disk usage in percentage shows how much of the allocated disk space is currently in use. Once a

target threshold is reached (50 – 90 %), oldest processed messages are deleted.• Disk usage is below target if there is plenty of free disk space.

• Disk usage is at target during normal operation; processed messages are deleted as required.

• When disk usage is above target, pending messages fill the fax storage beyond the threshold. Noprocessed messages are kept.

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Inbound MessagesThis folder stores all fax messages inbound to FoIP storage.• The Pending subfolder shows inbound messages which have not yet been delivered to an application.• The Processed subfolder shows messages which have been confirmed as received by an application.• The Processed | Failures subfolder shows messages confirmed by an application where a failure was

encountered during fax reception.

Information about inbound fax messages in the storage are organized into columns.

Column Description

TSI The fax number of the calling subscriber.

Pages Number of pages of the fax message.

Size Size of the fax message.

Reception Start Time Start time of receiving messages.

Reception End Time End time of receiving messages.

Error The error category displays „OK“ or a simple error cause. See Error Codes.

Code A more detailed error code. See Error Codes.

Diagnostics Additional information about the error.

Info For inbound fax messages, the information column can take the following values:• Locked: Fax is being delivered (available for unconfirmed messages).• <empty>: Fax is neither locked nor deleted.

Commands A command column offers the following options:

Click to display the content of the fax in a TIFF viewer or to save thefax in TIFF format.

Click to display additional information about a fax message. See thefollowing figure (Outbound Fax Properties) for additional details.

Click to view the source of the message in XML format.

Click to move the fax from the pending to the processed folder.

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Outbound MessagesThis folder stores all outbound fax messages.• The Pending subfolder lists messages which are queued for sending or are currently being sent.• The Processed subfolder shows all sent and unsent messages.• The Failures subfolder displays messages that could not be sent successfully.

Information about received fax messages in the storage are organized in several columns:

Column Description

Destination Destination fax number.

Pages Number of pages of the fax message.

Size Size of the fax message.

Send Time Date and time of sending.

Error The error category displays „OK“ or a simple error cause. See Error Codes.

Level The connection level defines the maximum connection level that could be reached withinthe call

Code A more detailed error code. See Error Codes.

Diagnostics Additional information about the error.

Retries Number of retry attempts left for sending out the fax (relevant for queued messages).

Info For outgoing fax messages, the information column can take the following values:• Locked: fax is being sent.• <empty>: fax is queued for sending.

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Column Description

Commands A command column offers the following options:

Click to display the content of the fax including/without header line in aTIFF viewer or to save the fax in TIFF format.

Click to display additional information about a fax message. See thefollowing figure (Outbound Fax Properties) for additional details.

Click to view the source of the message in XML format.

Click to move the fax from the pending to the processed folder.

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Chapter 4

Integration with KCS

This chapter describes the specific configuration details for connecting with Kofax Communication Server(TCOSS) and how to use Kofax Communication Server Fax over IP with the Kofax Communication Server(TCOSS).

You must have knowledge about TCOSS and how to use TCOSS with TC/FoIP v2 (XCAPI).

The following topics are covered in this chapter:• Installation of fax, voice, and message waiting functionality• Supplementary services• Special configuration options, such as failover• UFI channel configuration, tracing options

Installation of Fax for KCSPerform the following steps to install Fax for KCS.

1. Step 1: Set Up TCOSS Server for FoIP

2. Step 2: Set Up FoIP

3. Step 3: Verify Installation

4. Step 4: Set Up Call Peers

Step 1: Set Up TCOSS Server for FoIPPerform the following steps on the TCOSS server.

Note We assume that TCOSS has already been installed.

1. Configure User Module for FoIP (UFI) channels on the TCOSS server as described in the TCOSSFax over IP and TCOSS Configuration Manual.

2. Use the License Maintenance tool to add Fax over IP Channel (T.38) licenses on the TCOSS server.3. Restart TCOSS.

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4. Verify that the following registry values (key = key HKLM\Software\TOPCALL\TCOSS\LINK\LANxxx) are correctly configured:

a. IpToConnect must be set to the host name or IP address where Kofax Communication ServerFax over IP has been installed.

b. Password may optionally be checked by Kofax Communication Server Fax over IP.Refer to the Line Server Model 305 Manual or TCOSS Configuration Manual ("BranchboxConfiguration") for further details.

Step 2: Set Up FoIPInstall Kofax Communication Server Fax over IP with default configuration.

It uses 127.0.0.1 (localhost) as gateway address and therefore does not require any FoIP environment.

Step 3: Verify InstallationThis step verifies that steps 1 and 2 were performed correctly.

1. After about 1 minute, TCOSS should be connected with Kofax Communication Server Fax over IP.You can verify this connection by using the KCS Monitor.The “Connector to TCOSS 7” info line must show the same percentage of disk usage as the info linein TCOSS.

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2. Send a fax to any UFI channel.Since the default configuration is set to loop-back, the fax must be received by any other channelwhere the dialed number is received as inbound number on receiving side. Activity can be monitoredwith KCS Monitor (Message Server view). An example is shown in the following figure.

It is also visible in the FoIP info line:

Step 4: Set Up Call Peers1. From the Start menu, select Kofax Communication Server | Configure FoIP and configure call

peers as described in List of Call Peers.2. Save the configuration and exit the configuration tool. Restart FOIP using KCS Monitor.3. Check if fax reception and transmission over the configured call peer is working.

Installation of Message Wait for KCSKCS FoIP supports two different types of message wait integrations:• MWI control number (used by Cisco CallManager / Cisco Unified Communication Manager): Message

wait is handled by dialing a specified extension for message wait on/off.• Sip Notify according to RFC 3842 (used by Siemens OpenScape Voice): Message wait is handled by a

special SIP message (NOTIFY).

This section provides step-by-step instructions on how to install a message waiting solution with CiscoCallManager based on KCS FoIP. See Message Wait via SIP Notify (RFC 3842) for integration withSiemens OpenScape.

This installation assumes that the steps listed in Installation of Fax for KCS have been completed. Fordetails, refer to "Messages Wait Integration" of the VoIP / FoIP Environment Guide.

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Step 1: Configure KCS FoIPLaunch KCS FoIP Configuration Utility. In the “KCS Integration” section, set MessageWait to “MWI controlnumber (Cisco CM, UCM)”.

Step 2: Configure CallManager and Test Message Waiting1. Create two special unique extension numbers within each Cisco CallManager cluster.

Refer to the VoIP / FoIP Environment Guide for instructions. In this example, we assume “4632” (off)and “4631” (on).

2. Start the TCfW Communication Server Client and send blank messages to the following address:free,F:TCMWION;4632;4630

The message waiting indication lamp on the telephone number 4630 will be lit.free,F:TCMWIOFF;4631;4630The message waiting indication lamp on the telephone number 4630 will fade.In the examples above, “F:” is a channel group, “TCMWION” and “TCMWIOFF” are case-sensitivecommands, “4632” and “4631” are the numbers defined in Cisco CallManager for message waitindication, and “4630” is the number of a telephone station.

Step 3: Create Services and Events in TCfW1. In TCfW, from the Admin menu, select Services.2. Create the MWION and MWIOFF services as follows:

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3. In TCfW, from the Admin menu, select User Profiles. For the user with telephone extension 4630,create the following events:

Message Wait via SIP Notify (RFC 3842)When using SIP Notify (such as with Siemens OpenScape Voice), the following changes from the aboveinstructions are required:

1. Configure Message wait mode to “RFC 3842” instead of “MWI control number”.

2. Use the mailbox access number instead of the MWION/MWOFF in the test call (Step 2) and MWION/MWIOFF services (Step 3).

Installation of Voice for KCSFoIPv3 may be integrated with the KCS Voice Server in addition to the integration with TCOSS. Theconfiguration steps to use voice with FoIPv3 are the same as with Line Server Model 305.

This chapter describes an installation example involving TCOSS, Voice Server and KCS FoIP v3. Thefollowing is assumed:• TCOSS server, Voice Server and FoIPv3 and TC/Player are running on the same machine• Fax functionality and the call peer to a Gateway (or similar) is already configured (as described in

Installation of Fax for KCS).• Incoming calls to extensions 7190 .. 7199 are routed to FoIPv3.

The goal of the example it to set up a simple Voice Access solution where messages can be recordedwith extension 7198. Extension 7199 can be used to access the recorded messages. All other extensions(7190 .. 7197) should be used for fax reception. It should be possible to play voice messages via TC/Player on the telephone. The example does not rely on the availability of call-diversion features, whichshould be used in a real productive environment to optimize a voice access installation.

Refer to the Voice Server documentation for further information about all voice related features.

Install Voice Server1. Start the Server Package setup. From the TC/VoiceAccess (Group), select the TC/VoiceAccess

application.

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2. Set the server path and the access number. The default password for user TCVMAIL (which isinstalled below) is TCVMAIL.

3. On the next screen, use the default values.

4. Select “LS1 or FoIP” as the telephone system.

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5. Enter IP address and password of FoIPv3. In this example, we use 127.0.0.1 because it is runninglocally.

6. After the installation, when TCOSS has been started, run C:\TCOSS\System\vmaildep.bat toinstall the required services and users.

7. For a more detailed description of various setup options, refer to the Voice Server documentation.

Configure Voice Support in FoIPBy default, the connection the KCS voice service is disabled in KCS FoIP.

To support the connection to the voice server, do the following:1. Start the FoIP configuration utility.2. Verify that the Reception Mode is set to “TCOSS”. Verify dial peer settings.3. To enable voice support, in the KCS Integration tab, select EnabledVoiceServer.

4. Restart FOIP using KCS Monitor.

Enable Voice in the UFI ChannelsBy default, voice support is disabled in the UFI channels.

To enable voice support, perform the following steps.1. Enable channel for outbound voice calls by setting position 1 of configuration line 295 to “01”.

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2. Enable channel for inbound voice calls by changing the number conversion table (lines 254 to 283)so that all received numbers that should be handled by voice are converted into a number whichstarts with “V”.Example config line 254 ‘17908=V7908 ,config line 254 ‘17908=V7908 ,

config line 255 ‘17909=V7909 ,

Test the First Incoming Voice CallVerify that TCOSS, Voice Server, and FoIP are running and connected to each other.

1. Open the FoIP status monitor and verify the connections to TCOSS and Voice Server.

2. Check the state of TCOSS and TCECP (Voice Server) in KCS Monitor.

3. Call the TC/VoiceAccess access number (in our example, 7199) with a telephone. You should heara prompt saying “This is TOPCALL voice mail, please enter the number of your mailbox and presshash”. Additionally, you should see activity in the Calls line of FoIP application in KCS Monitor:

Leave a Messages via TelephoneTo record a message to extension 7198, perform the following steps.

1. Create a KCS user with an inactive address (service=VOICE, number=7198).2. Call number 7198. You should hear the message “The person with extension 7198 cannot answer

your call. Please leave a message after the tone”. Now leave your message.3. Call number 7199. Type “7198” as the mailbox number and an empty password to play to recorded

message.

Test Outgoing Voice Call with TC/Player1. Install TC/Player from KCS Client Applications and open the player.2. Open any WAV or TCS file.

You can find sample wav files in the directory C:\topcall\SHARED\voicelib\TCVMail\01.

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3. Play the file via local sound card - select Tools | Use Soundcard and click Play.If you do not have a sound card, skip this step.

4. Configure the connection to your Voice Server.An example for a local connection is shown below.

5. In the TC Voice Player tab - set the telephone number where messages should be played.6. In the Voice Server tab, set the name/IP address of the voice server.

7. Play the file again via telephone - select Tools | Use Telephone and click Play.Your telephone should ring and the message should be played.

Unique Single Number SupportThe idea behind USN is to provide only one number for telephony (voice) and fax. This offers costreductions for the company in countries where you have to pay for each extra digit in your telephonenumber.

This section provides some hints and FoIP specific details for using USN with FoIP. It is assumed that youare familiar with USN as described in the Voice Platform Technical Manual.

Switch to Fax by AttendentThe Attendant is a feature of TC/VoiceAccess to provide the caller a voice menu with a list of options. Ifthe Attendant is configured and USN is enabled (registry key TCVMAIL\doUSN is set to 1) the caller canpress "9" to leave a fax for the called user.

Switch to Fax by Voice ScriptThe voice script may set the connection property "faxcall" in order to transfer control of the current call toFax. The assigned number (for example, digits received by any IRV dialog) is used for inbound routing. Ifthe assigned number is empty, the fax uses the original received extension for inbound routing.

The corresponding Voice XML element is:

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<transfer name="transfer1" destexpr="'tel:' + extension" bridge="true">

Refer to the "Transfer the Voice Call to Fax" section in the Voice Platform Technical Manual.

Switch to Fax Due to Fax CNG Tone DetectionThe automatic fax detection is disabled by default. It can be enabled in position 4 of configuration line 295by setting the fax detection duration in seconds.

If enabled, the call is transferred from Voice mode to Fax upon detection of CNG within the configured faxdetection duration. The CNG signal is ignored if the detection is disabled (or after the detection time-out).

The detection of CNG is supported by using RFC 2833 (RTP-NTE). Other options, such as detecting CNGtone from received G.611 samples are not supported.

Switch to Fax Due to T.38 Mode Activation by the GatewayDepending on the gateway configuration, the gateway may request a mode change from G.711 to T.38upon detection of CNG. This case is handled by the following rules:• If the fax inbound mode is set to 60 (Use G.711 pass-through mode and prevent switch to T.38), the call

control rejects the mode change. The call is disconnect if reject fails.• If the automatic fax detection is enabled in position 4 of configuration line 295, the mode is switched to

fax. Otherwise the call is disconnected.

HintsAs with Line Server Model 305, all numbers with voice calls are converted as with fax calls using both thenumber conversion table. Optionally, the configured PABX prefix may be used (see configuration lines 131to 133, 2nd position of line 295).

Supplementary ServicesThis section describes supplementary services, such as call diversion, DTMF reception, call transfer andmore.

Call Diversion (Redirecting Number)Call diversion (or call forwarding) is a feature on some telephone networks that allows an incoming callto a called party to be redirected to other telephone number. There is a number of conditions that cantrigger call forwarding, such as called party busy, not answering, unconditional forwarding, and more. Calldiversion is also used to forward calls to the voice mail.

During call diversion for voice mail, the original called number must be delivered to the Voice Server inthe signaling protocol. When using multiple redirection, KCS FoIP may receive two or more divertingnumbers. In such a case, the configuration value Call Diversion Mode in the KCS Integration sectionselects either the original called number (number which initiated the call) of the last redirecting number(number which redirected the call to FoIPv3).

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Via SIPThe SIP protocol uses a Diversion header field. If you use a Cisco CallManager you need to enablethe delivery of this header field. Open the configuration page of your SIP trunk and enable RedirectingDiversion Header Delivery for inbound and outbound.

Via H.323KCS FoIP supports the following alternatives to deliver the original called number via H.323:• H.450.3: Delivers the original called number and diverting number (last redirecting number)• Q.931: Delivers one number• Cisco CallManager proprietary: Delivers the original called number in a non-standard proprietary

field. The delivery of this number must be enabled at the Cisco CallManager• Tunneled Signaling Message: Gateway can transmit the Setup message received via ISDN

encapsulated in a tunneledSignallingMessage.

Refer to the VoIP / FoIP Environment Guide or PBX documentation for further details.

Suppression of Redirecting NumberKCS FoIP can be configured to suppress the redirecting number; that is, the number of the party that hasdiverted the call is not evaluated. This function can be used where you want to use a common voice mailbox for multiple users.

To suppress the redirecting number, do the following:1. Using WConfig, for UFI channels, set the most significant bit of line 295, position 3 to 1. For example,

if the current number was “01”, change it to “81” hexadecimal.2. Enter a similar line to the number conversion lines of the UFI channels (starting with line 254):

Lxxxx=Vxxxx.Where xxxx is the number where most of the phones are redirected to, for example 5960.

3. Restart FoIP.Example Assuming that the feature is active and number conversion line “L5960=V5960” has beenset:• If a VoIP call is redirected to the FoIP component and its called party number equals 5960, the

voice call is routed into the voicemail box determined by the redirecting number arrived along withthe call setup.

• If a VoIP call is redirected to the FoIP component, but its called party number doesn’t equal 5960,the voice call is routed into the voicemail box determined by this called party number and thepossibly received redirecting number is suppressed.

Use Redirecting Number for Fax CallsBy default, inbound fax messages can be distributed according to the called party number (DDI/MSN).

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To use redirecting number for fax calls, do the following:1. Using WConfig, for UFI channels, set the least significant bit of the line 286, position 8 to 1 (for

example, from “00” to “01”).2. Enter similar lines to the number conversion lines of the UFI channels (starting with line 254):

ATN~=0~ATI~=00~ATS~=~

Note PBX might send the redirecting number along with the flag national/international/subscribernumber. This flag must be converted to the corresponding prefix.

3. Restart FoIP.If this feature is enabled and a redirecting number is present, the redirecting number is used insteadof the called party number.This feature does not work if the redirecting number is suppressed. See Suppression of RedirectingNumber.

DTMF ReceptionVoIP detects DTMF signals delivered via RTP according to RFC 2833 (RTP-NTE). This signaling canbe used with SIP and H.323, where support of RTP-NTE for Sip has to be enabled in the configuration.Furthermore, when using H.323, DTMF signaling according to H.245 (alphanumeric and signal) is alsodetected. DTMF signaling using SIP notify is not supported.

DTMF Generator for Two-Stage Outbound DialingIt is possible the make an outbound fax call with additional DTMF digits or pause characters after theconnection has been established. The syntax is:

N=...<Number> [‘T’<Dtmf>] ['-'<answerback>]

Where <Dtmf> may contain any valid DMTF digits (0..9, #, *, A, B, C, D), P (1s delay) or Q (4s delay). Youmay also use DTMF digits in the number to dial field of the web GUI test page.

The DTMF signals can be generated as G.711 or RFC 2833 (RTP named telephone events) as describedin the FoIP configuration section “Fax”.

The DMTF digits are generated using the following timing.

Timing parameter Value

Delay before 1st digit (time between connect and firsttransmitted digit)

>=50ms

Duration of DTMF digits 100ms

Delay between DTMF digits 100ms

Restrictions:• The transmission of DMTF signals or delays is not support for outgoing voice calls.• The transmission of DTMF signals via H.323 (using H.245) or SIP (using SIP Notify) is not supported.

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• The transmission of DTMF signals as G.711 audio and RFC 2833 telephone event as RTP redundany(according to RFC 2198) is not supported.

Call TransferCall transfer is a feature on some telephone networks that allows a party involved in a particular call(the “transferor”) to transfer another party of this call (“transferee”) to a new destination number (the“transferred-to-party”). For example, imagine a caller who is redirected per call diversion to the KCS Voiceserver and elects to be connected to the operator instead of leaving a voice mail to the originally calledperson: behind the scenes, the KCS Voice server acts as the transferor, the caller is the transferee andthe operator is the transferred-to-party. After the call transfer is completed, the transferee (the caller) isconnected with the transferred-to-party (operator).

For information about call transfer configuration in KCS FoIP, see KCS Integration.

KCS recognizes two basic call transfer variants:• External Call Transfer: The transferee is connected with the transferred-to-party through telephony

infrastructure (the PBX) and the original call towards KCS Server is disconnected• Internal Call Transfer: The transferee is connected with the transferred-to-party through KCS Server

occupying two KCS connections during the entire call

It is up to particular voice application/function to determine which call transfer variant is the most suitable:for example, the call sender function by default uses the internal call transfer and the attendant functionthe external one.

For more information about internal / external call transfer variants and configuration in voice components,refer to the Voice Platform Technical Manual.

Internal call transfer uses the outgoing number conversion table (FoIP channel configuration). Thesupported protocols are SIP and H.323.

External call transfer does not use outgoing number conversion table. Only SIP is supported.

In the case of H.323, external call transfer requests automatically fall back to the internal call transfer.

For sample configuration with Siemens HiPath 8000 and Cisco CUCM 7.x PBXs, refer to the VoIP / FoIPEnvironment Guide.

Note When using call transfer with voice mail, we recommend that you enable only a single G.711codec. Having both enabled can occasionally lead to payload mismatch. See Voice.

Configuration HintsThis chapter describes hints for some typical use cases.

Setting Own Calling Party NumberFor outgoing calls depending on the used VoIP/FoIP system or configuration, it may be required to set thecalling party number for outgoing calls.

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For example, when using a Cisco gateway, the calling party number can be used the select the dial-peerwhich is applied for the outgoing call. If no dial-peer matches, the global gateway configuration is applied(defined in voice service VoIP and others). If the global configuration differs from the FoIP configuration,problems can occur; therefore, it is highly recommended to set the calling party number for outgoing calls.

The calling party number can be set fixed for the channel (with a number conversion rule) or userdependent (with a ++CID line in the users template or by the cost center parameter and an entry in Arr99).Refer to the TCOSS System Manual for details.

This is an example for setting the calling party number to 8123 for all outgoing calls in the TCOSS channelconfiguration. It is recommended to use configuration lines 276 to 279 which already contains an outgoingnumber conversion entries in the standard configuration as shown in the example below:'8*~=8123%00~ , 276 (normalized number)'80~=8123%0~ , 277 (national or internal number)'8I~=8123%I~ , 278 (internal number)'8~=8123%~ , 279 (local number)

The example above works with FoIPv3 but not with FoIPv2 or LS1, because FoIPv2/LS1 required that thecalled party number type (prefix “I”, “TI”, “TN”, “TC”, …) must be at the begin of the converted line (and notbefore the called party number). An alternative configuration which is supported by FoIPv2, FoIPv3 andLS1 is shown below:'8T?~=T?8123%~ , 275 (control calls; with number type)'8*~=8123%00~ , 276 (normalized number)'80~=8123%0~ , 277 (national or internal number)'8I~=I8123%~ , 278 (internal number)'8~=8123%~ , 279 (local number)

The additional conversion line 275 is required, otherwise message wait control commands do not work.For example, TCMWION;4632;4630 would be converted to 8123%TCMWION;4632;4630 which doesnot work with LS1 because it does not start with “TC”. But a conversion to TC8123%MWION;4632;4630works.

Fax on DemandFax on demand is supported as described in the TCOSS System Manual for Line Server Model 305.

High Availability SupportThe following features are used in order to support highly available FoIP solutions:• TCOSS LANLINK may connect to an alternative IP address.• FoIPv3 supports failover or load balancing for outgoing calls with up to 8 different call peers (each call

peer represents any IP telephone system host/port).• FoIPv3 supports a call-peer check that can be used to prevent a FoIP instance without working

connection to the telephone system to get a connection from TCOSS.• Inbound failover and load balancing is typically provided by the IP telephone system. If your IP

telephone system does not support it satisfactorily, you can use a multicast address for inbound callsas described in Inbound SIP Failover with Multicast Addresses. The Behavior If an Incoming Call IsRejected section describes the behavior when FoIP is running but an incoming call cannot be accepted.

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Reference ConfigurationThis reference configuration uses KCS Tandem server, 2 FoIP instances and 2 gateways. Solid line isused for components and connections that are used during normal operation. Dashed line is used forcomponents and modules that are used as fallback only.

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Note• The IP addresses are just examples. You can choose any address.• The assumption is that TCOSS and FoIP are running on different machines. You can also apply

this example when running FoIP on the TCOSS server. For better distribution CPU utilization duringnormal operation, it is recommended to run FoIP #1 on the secondary master.

• If you have a single TCOSS system, ignore the Secondary Master in the drawing above.• If you have an IP system with a single IP address, ignore Gateway #2 in the drawing above.

Configuration hints

• All UFI channels use the same LANLINK. The IpToConnect registry value of the LANLINK has to be setto "10.0.1.1 | 10.0.1.2" both on the primary and secondary master.

• All call peer failover strategies (configuration line 291, position 12, in all UFI channels) are supported,but the load balancing strategy is assumed and recommended.

• Both FoIP instances must have call peers to both gateways. See sample figure below:

• The Call-Peer check mode should be set to “Check if any Call-peer is OK” in the FoIP KCSconfiguration. When using FoIP without fix of SPR70426, that is, 3.12.01 or older, the call-peer checkshould not be used with multiple call peer configurations.

• Both gateways should configure FoIP #1 as primary IP and FoIP #2 as secondary IP. Note that thisconfiguration step depends on the used gateway or IP BAPX. Refer to the VoIP / FoIP EnvironmentGuide for further details. An example for Cisco gateway can be found in Call-Peer Check.

Normal operation state

The primary master is connected with FoIP #1. FoIP #1 distributes outgoing calls (using round robin) togateway #1 and gateway #2. Incoming calls received at any gateway are handled by FoIP #1. Both thesecondary master and FoIP #2 are in a standby mode.

Failover scenarios

• Secondary master, FoIP #2, and link connections 2, 3, 4, 7, and 8 fail: This failures just removedfault tolerance because all components are not used during normal operation.

• Primary master fails: The secondary master will restart stand-alone and connect to FoIP #1.• Primary master and FoIP #1 fails: The secondary master will restart stand-alone and connect to FoIP

#2.• FoIP #1 fails: The primary master connects to FoIP #2.• Gateway #1/#2 fails: Outbound failover automatically redirects all calls to gateway #2/#1.• Network connection #1 fails: The primary master connects through connection #2 to FoIP #2. The

gateways may still try to signal incoming calls to FoIP #1 which returns an appropriate error code asdescribed in Behavior If an Incoming Call Is Rejected. The gateway will reconnect the incoming call toFoIP #2.

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• Network connection #5 fails: All outgoing and incomings calls are done via gateway #2. Incomingcalls on gateway #1 fail because the primary IP is not reachable and FoIP #2 does not have aconnection to TCOSS. It is the responsibility of the connected telephone system that incoming calls togateway #1 are retried at gateway #2.

• Network connection #6 fails: Similar to failed connection #5, but all calls are done via gateway #1.• Network connections #5 and #6 fail: The call-peer check in FoIP #1 detects that no call peer is

available. FoIP #1 disconnects from TCOSS. TCOSS LANLINK L.1 reconnects with FoIP #2 which hasa working connection to both gateways.

Outbound Failover StrategiesWhen used with TCOSS, Kofax Communication Server Fax over IP supports outbound failoverconfiguration. You can set up a maximum of 8 call peers (as described in chapter List of Call Peers).When a call peer is unavailable, another will be used automatically. Should a call peer go down (or ifnetwork is busy or the destination has no free channels), FoIP attempts to reconnect using the next callpeer, and others. The call peers may even use different protocols. For example, the first call peer can beusing the H.323 protocol and the second SIP.

KCS FoIP supports multiple failover strategies:• Use first available call peer: Outgoing messages are sent out by the call peer configured in the first

position of the call peer table. If it fails, the next call peer is attempted. This continues until a call peercan handle the call or all call peers were attempted. The next call starts again with the first call peer.

• Use current call peer until it fails: Messages are sent out by the call peer that was most recentlysuccessfully used. If that fails, other call peers are used.

• Load balancing (round robin): Call peers are used sequentially; each message is sent by another callpeer. This strategy also takes into account the last known state of a call peer; unavailable call peersare skipped, until a certain time period passes (30 seconds for busy and 5 minutes for unreachable).However, if all call peers are marked unavailable, FoIP will attempt to use them for sending anyway.

Failover strategies are configured via WConfig in config line 291, position 12. See Description of UFISpecific Configuration Lines.

For each TCOSS channel, it is possible to configure which call peers are available for outgoing calls. Thisis configured via WConfig in config line 291, position 9. Position 10 determines the call peers for incomingcalls. See Description of UFI Specific Configuration Lines for details.

Failover is currently not supported for fax messages sent via storage.

Call-Peer CheckThe call-peer check may be activated in the KCS section of the FoIP configuration. It prevents situationswhen FoIP gets a LANLINK connection from TCOSS without a working connection to the IP telephone

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system. In that case an event log entry (ID 35013) will be created and the following alert is shown theFoIP status overview:

The check sends a SIP OPTIONS request to all active SIP call peers that should be checked. Theconnection to TCOSS is allowed if at least one checked call peer returns a SIP response value between200 and 499 or if no call-peer should be checked.

After FoIP is connected with TCOSS, the check is repeated every 60 seconds. If this check fails, FoIP willbe restarted with an event log (ID 35013 - Red alert 'FoIP will be stopped because the checked call-peersare down …)

Failover Gateway Configuration ExampleTwo dial-peers with the same MSN, one with the IP of the primary and one with the IP on the secondarymaster, must be configured on both gateways. This has only been tested with Cisco 2600 and 2800gateways.

Here is an example of such a dial-peer configuration for a Cisco 2620 gateway.dial-peer voice 80000 voipdestination-pattern 8...progress_ind setup enable 3session target ipv4:10.0.1.1dtmf-relay h245-alphanumericcodec g711alawfax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback noneip qos dscp cs5 mediaip qos dscp 31 signalingno vad

dial-peer voice 80001 voipdestination-pattern 8...progress_ind setup enable 3session target ipv4:10.0.1.2dtmf-relay h245-alphanumericcodec g711alawfax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback noneip qos dscp cs5 mediaip qos dscp 31 signalingno vad

On incoming calls (from ISDN to IP) the gateway matches the first dial peer by its destination patternto the incoming MSN. The dial peer 80000 will win. If the IP address is unreachable it will attempt thenext dial peer with the same destination pattern (80001) which leads to the secondary master. Pleaseconsider that the sequence of the dial peers in the gateway configuration is relevant for matching them.You can also use the preference parameter in the dial-peers to influence that (refer to the Cisco IOSdocumentation for details).

Inbound SIP Failover with Multicast AddressesFoIP may optionally receive inbound SIP calls from a multicast address. Multicast inbound provides loadbalance and failover for inbound calls without special handling in the IP telephone system.

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If multiple FoIP instances are active with the same multicast address, each one starts with the inboundcall (like a multi-ring feature). The first instance which accepts the call (by sending a 200 OK response tothe INVITE message) gets the call. This instance also informs all other instance to release it, which willend the call with TCOSS error code “JS” and internal error number 12722 in the FoIP trace.

The following restrictions should be considered when using multicast• If you configure a multicast IP, IPv6 support will be disabled for SIP.• Multicast works only if a local IP is specified in the VoIP/FoIP network settings.• The IP system must be able to correctly handle inbound calls to a multicast IP address. This is the case

if the system complies with the RFC 3261 (SIP).• In real life, multicast traffic is often not forwarded via routers. Technically this is possible; however, it is

often a company policy not to forward multicast traffic via routers. Further problems with IP multicastare caused by either a RPF (reverse path forwarding) failure or by TTL (time to live) settings. In such ascenario, it is necessary to install all FoIP instances in the same network segment as the gateway wheninbound SIP failover via multicast addresses is used.

Configuration Steps1. Prepare the installation by defining an IPv4 multicast address where the configured UDP port

(default: 5060) is not used by other applications.Remark: Multicast addresses are within the range 224.0.0.0 … 239.255.255.255. They are notconfigured in the network settings, and you cannot ping them.

2. Configure the gateway / IP telephone system by using Multicast IP address in the dial-peer (trunk)configuration.

3. Configure FoIP:

a. Set VoIP/Network | Local IP to the interface which is connected to the Gateway

b. Enter the multicast IP in the configuration value SIP | MulticastAddress .

c. By default, the notification to all other instances after a call has been taken is sent tothe configured multicast address. This works but causes an additional multicast packetin the network. Do optimize the installation it is recommend to enter the Unicast IPs ofall other FoIP instances using the same multicast IP in the configuration value SIP |MulticastPeerAddresses.

ExampleFoIP Configuration

Configuration parameter FoIP #1 FoIP#2

OIP/FoIP network | local IP 10.0.1.1 10.0.1.2

SIP | Multicast Address 225.1.1.1 225.1.1.1

SIP | MulitcastPeerAddresses 10.0.1.2 10.0.1.1

Inbound call

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• The gateway starts the inbound call with an INVITE message [1] to the multicast address 225.1.1.1which is received by both FoIP instances.

• We assume that both instances are able to handle the call but FoIP #1 sends its 200 OK response [2]faster than FoIP #2 [3]

• The first 200 OK response which arrives at the gateway establishes the dialog. It confirms the dialogwith an ACK [4]. Later responses from other instances [3] are ignored by the gateway.

• FoIP #1 gets the ACK [4] and then sends a multicast peer notification (which is a special ACK) to allother instances.

• FoIP #2 gets the multicast peer notification [5] and then releases the call without further messages.• All further communication is done via unicast IP addresses between FoIP #1 and the gateway.

Failover scenarios

• FoIP #1 or FoIP #2 is not active: The call is handled by the other instance.• FoIP #1 is not connected with TCOSS: The negative final response will be delayed by 12s. If FoIP #2

is able the handle the call within this period, it gets the dialog. Otherwise, the call fails.• FoIP instances do not have any free channel: If a channel gets ready within 10 seconds (either in

FoIP #1 or FoIP #2) the call will be handled. Otherwise, it will be rejected.

Handling of lost UDP packets

• Lost multicast INVITE [1] or response [2]: The gateway retries the INVITE packet until it gets aresponse (standard behavior as defined by RFC 3261).

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• Lost ACK [4]: FoIP retries the response [2] after about 30 seconds. Afterwards, the call is releasedwithout further messages.

• Lost multicast peer notification [5]: FoIP #2 retries transmission for about of its response [3] until

SIP Registrar and H.323 GatekeeperAn H.323 Gatekeeper is an optional component in an H.323 network. If it is present, it performs addresstranslation, bandwidth control, admission control, and zone management.

A SIP Registrar is a server that accepts REGISTER requests and places the information it receives inthose requests into the location service for the domain it handles.

ConfigurationTo configure a registrar or gatekeeper in FoIP, you need to know the IP address of the server, port number(not necessary if default is used), a user name and password for authentication to the server, and aregistered number. See List of Call Peers.

To verify that FoIP can connect to the server, start FoIP Web Portal and click Status. You should findsimilar lines in your status screen.

When configured incorrectly, the following or similar lines should be included in your status screen:

Alternatively, you can monitor the connection status via KCS Monitor:

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Restrictions• If you select an H.323 Gatekeeper (“H.323 with RAS”) you must not use H.323 with any other active call

peer. Thus, failover via H.323 is not possible if an H.323 Gatekeeper is configured. The assumption isthat fail-over is handled by the Gatekeeper in that case.

• Only one registered number is supported.

Behavior If an Incoming Call Is RejectedIf an incoming call will be rejected different call clearing causes are used according to the following table.

Situations H.323 Cause SIP Status code Typical signal generated bygateways

Originator IP not accepted 21 (Call Rejected) 403 Forbidden Busy tone

Inbound user does not exist 1 (Unallocated number) 404 Not Found Special Information Tone

No TCOSS channel available,TCOSS file structure full

17 (User Busy) 486 Busy Here Busy tone

No connection to TCOSS, Noreception channel configured

27 (destination out oforder)

504 Server Time-out Busy tone

Other unexpected internalproblems

63 (service or option notavailable, unspecified)

500 Internal ServerError

Busy tone

Note The Special Information Tone is a repeated series of signals with 950Hz, 1400Hz and 1800Hz sentin that order.

If reception into storage is configured, the incoming call is always accepted. If the received fax cannot besaved because the storage is full, the fax transmission will fail afterwards.

A SIP OPTIONS request is used by some IP Voice Systems to check if our application is ready (forincoming calls) without making a real fax/voice call. If FoIP gets a SIP OPTIONS request it returns theSIP status code shown above with the exception that the called party number is never checked. Thuswhenever FoIP is ready to handle any inbound calls it returns a success (200 OK) to an OPTIONSrequest.

FoIP and Internet Telephony Service ProvidersInternet telephony service providers (ITSP) offer telephone calls over internet. Some providers also offerT.38 support, that is, fax over IP. Kofax FoIP has been successfully tested in such an environment.

In this scenario, no fax server on the customer side is necessary. Instead, FoIP connects directly to aserver of the provider.

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FoIP configuration needs to be adapted. Any call peer must be configured as described below:

1. Select “SIP with registration” as the protocol of the call peer (usually).

2. Enter the IP address and port of the ITSP server.

3. Enter your user ID and password (generated by your ITSP provider).

4. Enter your telephone/fax number(s) (assigned by your ITSP provider).

Internet firewalls may sometimes block communication. Make sure that the following ports are notblocked:• UDP Port 5060 for SIP call control• UDP Ports for outgoing connections from any internal port within 10000 .. 10999 to any external port

within 1024 – 65535 for voice/T.38 connections

While providers usually offer elementary software for managing fax inbox and outbox (a software faxclient), Kofax FoIP solution has the benefit of being integrated with TCOSS (or other applications). Noadditional software has to be installed on end user computers.

V.34 FaxKCS FoIP supports the V.34 fax (ITU-T T.30 Annex F) through T.38 enabled fax relay according to ITU-TT.38 §10. The V.34 has several advantages compared to older standards, such as V.17:• Faster transmission speed, up to 33.6 kbits per second• Faster handshake• More reliable transmission

To use high speed fax with FoIP, the following requirements must be met:• Both originator and recipient must support V.34• Gateway(s) between them must support V.34 via T.38

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To enable V.34 fax with FoIP:

1. In the KCS FoIP configuration, section Fax

a. Select the V.34 enabled check box

b. The OutboundT38Mode must be set to 10 or 20 in order to support transmssion with V.34.

c. The InboundT38Mode must be set to 10 or 15 in order to support reception with V.34

2. In UFI channel configuration, line 55, position 2, set the maximum transmission speed to “N” (V.34,33600 bps) or use the send mode switch ENx, where x defines the actual used image compression(0=MH, 2=MR, 6=MMR)

3. a. In UFI channel configuration, line 143, position 2, set the maximum bit rate for reception to“05” (V.34, 33600 bps)

b. T.30 error correction mode (ECM) is mandatory for V.34 fax. If ECM is not enabled, V.27/29/17is used instead of V.34

c. V.34 fax is supported via SIP only. It is not supported via H.232 calls.

Note V.34 fax is not used for the Fax Polling mode (used with Fax-On-Demand feature)

Support Variable Gateway IP for Outgoing CallsBy default, an outgoing message from KCS Server (TCOSS) uses one or more call-peers according to thechannel configuration. In that case the IP address is used according to the call-peer configuration. This isthe default and recommended configuration which covers most installation requirements.

If you want to address many different gateways you can overwrite the IP address from the call-peerconfiguration on a call-by-call base with a prefix in the destination number. This can be done using thefollowing number syntax:

[<Send Options>][<gateway-IP>’!’][‘I’][<destination number>]

Examples:

Number (w/o channel) Meaning

1234 Send a number “1234”. Use the configured gateway(s)

127.0.0.1!123 Send to number “123”. Since 127.0.0.1 is the loop-back address, the fax will be receivedby local FoIP (on any other channel)

<N30>10.20.30.40!I40 Send to number “40” on gateway 10.20.30.40. use the prefix for internal calls andoperation mode “N30” (non-ECM, V.29/9600bps, MH coding)

The variable gateway IP features must be activated in position 11 of configuration line 291. If it is disabled(which is the default), the specified gateway IP is removed from the number field but it is not used toestablish the outgoing call.

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Note The direct gateway IP address cannot be specified by the user if the default service FAX isused, because it uses the address type fax which removes the dots from the IP address (the numberconversion for outgoing fax numbers are applied; refer to the "Fax Number Conversion" section in theTCOSS Application Module Manual for more details). Instead, either a service with number type FREEmay be used or the gateway IP address will be added during rr99 routing.

Support of Static NATNetwork address translation (NAT) is used to hide addresses of private networks from other networks. Themost common use is a NAT router that connects a private network with the public internet.

The default behavior of NAT routers is that they support connections initiated by any host within theprivate network to any host in the public network but they block connections from the public network toany host in the private network.

If you need to access a private host from the public network, you can set up a static NAT which is a simpleone-to-one mapping of a private to a public address that works fine for many applications.

IP telephone calls via H.323/SIP are more complex because they are using different connections forsignaling and media data, where the signaling connection is used to exchange the media addresses. Inthat case the public member must get the mapped public (instead of the private) address of the privatemember. This can be done by using one the following approaches:• The NAT router is a proxy server. In acts as a man in the middle that understands the signaling protocol

(H.323/SIP) and correctly maps the internal addresses to external addresses. This is the recommendedoption. It does not require special configuration steps in FoIP.

• The application, such as FoIP, in the private network transports its public addresses (instead of the realused private) in the signaling messages. This option is called NAT traversal.

FoIP supports NAT traversal with H.323 using a single fixed public IP as described in the example below:

In network A (private), FoIP hasthe IP address 10.0.0.10. Set theparameter NatTraversalIp to itsexternal address 192.1.1.10. Use193.1.2.3 as Gateway IP.

NAT router links the internal address10.0.0.10 to external address192.1.1.10 using a static NAT rule.

In network B (public), gatewayconnects to FoIP using the address192.1.1.10.

To make static NAT work with FoIP, do the following:

1. In KCS FoIP configuration, H.323 tab, configure the value NatTraversalIp. See H.323 Signaling formore details.

2. On the router, map the internal IP address of FoIP to an externally accessible IP address.Gateway will communicate with FoIP using its external IP address.

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Remarks:• Depending in the used environment, you may optionally have to select the H.245 Tunneling option in

the H.323 Signaling tab.• NAT traversal is not supported with SIP. You must use a SIP proxy instead.

Endpoint NameThe endpoint name is transferred as display name for inbound and outbound calls. Depending on theused environment this name may be shown on the end-user’s telephone.

With H.323, a single endpoint name is used for all calls and all call-peers. It is taken from first active H.323call-peer where the User ID is not empty. If all H.323 User IDs are empty, the current windows user ID(such as “System” for default installations as “Local System”) is used as endpoint name.

Note Note For compatibility with future releases and easier understanding it is recommended to use thesame User ID for all active H.323 call-peers.

The endpoint name never transmitted with SIP.

T.38 vs. G.711 Pass Through FaxKCS FoIP supports fax via T.38 and pass through mode. A general overview of both modes can be foundin the figure below:

When using G.711 pass through instead of T.38 the fax modem function which converts between binarydata and audio signals moves from the IP infrastructure (such as gateway) into our application.

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T.38 is the preferred mode, but G.711 pass through mode is recommended/required in the followingcases:• T.38 is not or insufficiently supported by the IP environment.• Encryption of media data (using SRTP) is required.

Configuration details can be found in description of the configuration section Fax.

Note Silence suppression is not supported with G.711 pass through mode.

T.38 RedundancyT.38 redundancy is a forward error correction method that improves success rate if T.38 packets arelost in the network. This is done by adding the content of the previous T.38 packets as redundancy. Asa consequence, the size of T.38 UDP packets (and the bandwidth) increases with redundancy. Seeexamples below:

Assume that T.38 packets 4, 8, 9 are lost by the network:• With redundancy 0, the information about all lost packets (4, 8, 9) is missing. This missing information is

handled like a bad line condition; these are typically handled by the T.30 error correction mode.• Redundancy 1 is able to recover packets 4 and 9 but the size of each UDP packet is doubled. The

information about packet 8 is still missing.• Redundancy 2 is able to recover packets 4, 8, and 9 but the size of each UDP packet is tripled. The

information for all packets is available.

Note T.38 redundancy is a trade-off between the maximum supported packet-gap and the requiredbandwidth!

The generation of redundancy is configured in each T.38 endpoint. For example, if you want to handlepacket loss from your Gateway to FoIP, redundancy must be configured in the Gateway. It is not requiredto use the same redundancy values on both endpoints of a T.38 session. It is also not required to enableprocessing of the redundancy on the received side because the receiver automatically detects and usesredundancy.

T.38 endpoints without support of redundancy (like KCS FoIP before KCS 9.2) silently ignore receivedredundancy. This means that it is compatible with an endpoint using redundancy but it cannot recover lostpackets.

KCS FoIP supports generation and reception of T.38 redundancy information according to ITU T.38chapter 9.1.4.1. The generation of redundancy can be configured individually for low-speed (V.21 andnon-data events) and high-speed (V.27, V.29, V.17, V.34) T.38 packets. The required network bandwidthwith redundancy for some typical values are shown in the following table.

Configured redundancy Required network bandwidth during T.38 mode

Low-Speed High-Speed V.17 Fax V.34 Fax

0 0 30kb/s 60kb/s

1 0 30kb/s 60kb/s

3 1 45kb/s 90kb/s

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Encryption of VoIP/FoIP Network TrafficKCS FoIP supports the encryption of VoIP/FoIP network with SIP signaling as described in this section.Encryption is not supported with H.323.

Note This section describes the general behavior as it would appear if you are using a connection fromFoIP to FoIP. Refer to the VoIP / FoIP Environment Guide for 3rd party specific details.

Enable Transport Layer Security (TLS) for SIPTLS uses a key and certificate(s) to encrypt the SIP messages and to check the authenticity of the remoteside as it is done with https. The major difference to web browsing with https is that SIP connections maybe initiated from both sides so that each end of the SIP trunk has to support both the TLS server and theTLS client role.

The TLS server requires at least a valid key and certificate as shown in the following figure.

The SSL chain certificate is optionally required if the SSL Certificate was not directly generated by acertification authority that is known is trusted CA on the remote side.

If valid keys/certificates are installed on both sides of the trunk you can switch the SIP transport from UDP(or TCP) to TLS with the following changes:• Ensure that TLS is part of SipEnabledTransports and that a valid local TLS port is used.• Switch SipOutgoingTransport to TLS and check/update to remote port in the used call peers.

An example configuration for SIP/TSL is shown below:

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KCS FoIP may optionally check if the server has a valid certificate by using the following configurationsteps:

1. Add the certificate of the remote side (or its certification authority) as SSL Trusted CA Certificates inthe SSL Certificate configurations section.

2. Select the CheckCertificate check box for Sip Signaling.

Note• SSL Certificates/Keys are issued with an expiry date. You have the renewal them before expiry if the

certificate check is enabled in the removed side.• If you do not use the certificate check, the communication is still encrypted but not protected against a

man-in-the-middle attack.• Consider that the private key (as well as the Lanlink password) is encrypted so that it cannot be used

on a different Windows computer. This means that you have to re-enter the password and privatekey if you reinstall FoIP on a Windows computer even if you restore the complete FoIP configuration(SolutionConfig.xml).

Example 1: Configuration with the Default SSL KeyKCS FoIP has a default key that is self-signed and uses an invalid host name. It may be used for looptests with FoIP or for some environments that accepts such a certificate. In such case, you can enableTLS as described in the previous chapter without need to configure a new key.

Note The default certificate does not protect the SIP connection against a man-in-the-middle attack.

Example 2: Generate Your Own Self-Signed CertificateA self-signed certificate is a valid SSL key/certificate pair that is issued by the KCS FoIP computer insteadof a trusted CA.

Refer to the TWS manual for the details on how to request a key/certificate for some well-known (andtrusted) certification authorities.

Step 1: Create a self-signed SSL key/certificate

Use a computer with any KCS (9.2 or later) server product.

1. Open an administrative command prompt.

2. Change directory to C:\topcall\shared

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3. Execute the following commands:C:\topcall\SHARED>set OPENSSL_CONF=openssl.cnf

C:\topcall\SHARED>openssl req -config openssl.cnf -new -nodes -keyout private.pem -out request.pem -days 3650Loading 'screen' into random state - doneGenerating a 1024 bit RSA private key............++++++..........++++++writing new private key to 'private.pem'-----You are about to be asked to enter information that will be incorporated into your certificate request.What you are about to enter is what is called a Distinguished Name or a DN.There are quite a few fields but you can leave some blankFor some fields there will be a default value,If you enter '.', the field will be left blank.-----Country Name (2 letter code) [AU]:ATState or Province Name (full name) [Some-State]:AustriaLocality Name (such as city) []:ViennaOrganization Name (such as company) [Internet Widgits Pty Ltd]:KofaxOrganizational Unit Name (eg, section) []:Common Name (such as. server FQDN or YOUR name) []:foip.mycompany.comEmail Address []:

Please enter the following 'extra' attributes to be sent with your certificate requestA challenge password []:An optional company name []:

C:\topcall\SHARED>openssl x509 -req -in request.pem -signkey private.pem -out certificate.pem -days 3650Loading 'screen' into random state - doneSignature oksubject=/C=AT/ST=Austria/L=Vienna/O=Kofax/CN=at01d029.emea.kofax.comGetting Private key

C:\topcall\SHARED>

Note The common name must match with the host name as it is specified by the client. The parameter"-days" is used to define the validity of the generated key/certificate pair. All other input values are usedas additional information in the certificate only. The generated files private.pem and certificate.pem in C:\Topcall\shared are required in Step 2.

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Step 2: Install and test the SSL key/certificate

1. Open KCS FoIP configuration and copy the content of the certificate.pem file to configuration valueSSL Certificateand the content of the private.pem file to configuration value SSL Private Key.Leave the SSL Chain Certificate configuration value empty.

2. Select SSL Active in the HTTP tab.

3. Save the configuration and restart KCS FoIP.

4. Use a web browser (from any computer) and connect to the FoIP web portal using https://<host-name>:<port> where host-name must match (case insensitive) with „Common Name“defined in the certificate. The port depends on the HTTP configuration (default is 25081). You shouldget a warning that the certificate is not trusted. Ignore the warning and then the FoIP Web should beshown with a certificate error.

5. Click the Certificate error to view more details about the certificate.

Step 3: Optionally, install a self-signed certificate as trusted root certificate on the client

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If you are sure that your browser is connected to FoIP, you can also avoid the certificate warning byinstalling the certificate as trusted root certificate. This can be done (with IE 11) as described below:

1. Open Internet Options > Security > Trusted Sites > Sites. Add the URL of KCS FoIP as trustedweb site.

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2. Open the FoIP Web GUI and ignore the certificate warning. Click the certificate error to open thecertificate.

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3. The Certificate import wizards starts. If you want to make the trust for all users on this computer (forexample, also for services), select Local Machine as store location.

4. Select to install the certificate as a trusted root certification authority:

5. Complete the Wizard with defaults for all other steps.

6. Restart Internet Explorer and verify that the FoIP Web GUI opens without a certification error.

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Enable Encryption of Media DataMedia data encryption can be used for G.711 (Voice or Fax) only and only for SIP calls via TSL (asdescribed in the previous chapter). The encryption uses SRTP. To enable, set the configuration valueVoice > MediaSecurity to [2] Negotiate or [3] always.

If media encryption is not supported by the remote side,• Configuration option [2] falls back to G.711 without encryption.• Configuration option [3] causes a call setup failure.

Voice PromptsIf KCS FoIP is integrated with KCS incoming calls may be used for commands or incoming calls mayfail due to some number checks. In such a case KCS FoIP provides an audible feedback similar to LS1.But instead of using 1 beep for success and 3 beeps for errors, KCS FoIP tries to play a prompt file thatmatches the internal error code (as described in chapter 7.2.2) that causes call termination. If no matchingfile is found the file OK.wav (for success) or Error.wav (for failure) is played. All prompt files are installedin sub folder prompts (based on FoIP installation folder). An overview of the used KCS FoIP prompts isshown in the following table.

File Usage Content

42321.wav Mailbox command (71, 72 or 73) accepted “Command accepted”

42331.wav DTMF prompt 600ms/700Hz + 300ms/880Hz

42332.wav BER Test-loop prompt 1s/880Hz + 1s/700Hz

42410.wav Fallback if an error in the inbound number wasdetected that could not be mapped to errorreasons 42411-42415

“Bad inbound number unspecified”

42411.wav A fax server/routing command was entered withan incorrect syntax.

“Bad syntax of inbound number. For exampleinvalid command.”

42412.wav The specified inbound address does not exist “The specified inbound address does not exist”

42413.wav An invalid user (8xxx) command was entered “The specified user or password is not valid”

42414.wav A fax mailbox/routing command failed “A 7x command failed”

42415.wav The requested function is not allowed accordingthe security configuration in configuration line237/240.

“The requested function is not allowed withcurrent security configuration”

43103.wav Incoming voice call without connection to voiceserver

“Incoming call has been rejected because novoice server is connected.”

Error.wav Fall-back for error response 3 beeps

OK.wav Fall-back for successful response 1 beep

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User defined promptsYou can also replace the pre-installed voice by any user defined prompt, by creating a .wav file (8000samples/s, mono, G.711 A-law or G.711 u-law) with the same name in sub folder config\prompts.

If a prompt has to be played, KCS FoIP uses the prompts in config\prompts with a fallback to the pre-installed prompts in sub folder prompts. For example, if an inbound number does not exist, then the firstexisting file according the list below is used:• Config\prompts\42412.wav

• Config\prompts\Error.wav

• Prompts\Error.wav

Multiple FoIP InstancesSince KCS 10.0 it is possible to install up to 21 instances of FoIP via KCS setup. The instances use theregistry keys FOIP, FOIP01, FOIP02… FOIP20. Instance FOIP uses the installation location and registryvalues that was also used by prior versions of KCS FoIP. The new instances (FOIP01 – FOIP20) areinstalled but they are not automatically started by TCSRV in order to prevent port conflicts with FOIP.Whenever you add a new instance, the following steps must be done after KCS setup:• Use the KCS FoIP configuration tool and change all resources (such as local IP address or port) that

may conflict with any other instance.• Start the new instance manually with KCS Monitor. Check if startup succeeds without resource conflicts• Change registry value HKLM\Software\TOPCALL\FOIPxx\Autostart to 1 so that this instance will

be started automatically during next startup of TCSRV.

Configuration ValuesThis section describes the UFI channel configuration. It also describes the tracing possibilities in FoIP andlog entry filtering.

UFI Channel ConfigurationThe following table gives you an overview about the meaning of the UFI configuration files.

Config Line Position Meaning with UTF/UIF Meaning with UFI

1 – 50 TAM configuration Same as UTF/UIF

51 Own answer back Same as UTF/UIF

52 Default TCI/CSI Same as UTF/UIF

53 Back reception mode Same as UTF/UIF

54 Comment Same as UTF/UIF

55 Send options Same as UTF/UIF

56 Retransmission mode Same as UTF/UIF

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57 Send break page Reserved (UFI does not support sending abreak page)

58 Expansion of last page Same as UTF/UIF

59 Comment Same as UTF/UIF

60 – 65 Dialing parameter Reserved. KCS FoIP uses fixed parametersare DTMFs.

66 DC/DC converter Reserved

67 – 70 Local ring parameter Reserved

71 – 83 T.30 configuration Reserved

84 – 130 Call progress detection configuration Reserved

131 – 133 Automatic PABX prefix Same as UTF/UIF

134 Error counters Same as UTF/UIF

135 Output level Reserved (Output level depends onGateway)

136 – 142 Page layout for sending Same as UTF/UIF

1 Reception mode Same as UTF/UIF

2 Maximum bit rate for reception Same as UTF/UIF

143

3 Show transmission parameters in authorfield

Defines if enhanced error info or transmissionparameters should be written into the authorfield.

144 – 145 Received page defect limits Same as UTF/UIF

146 Fine mode reception Reserved (fine mode reception is alwaysenabled)

147 Date format Same as UTF/UIF

148 – 179 Code conversion table Same as UTF/UIF

180 Configuration of DTMF detector Reserved, DTMF digits must be detected bythe gateway (DTMF relay function)

181 – 230 Call control state machine Reserved

231 – 234 FXC Tracer configuration Same as UTF/UIF

235 – 240 Inbound prefix and Fax Server/Routerconfiguration

Same as UTF/UIF, but not that Server andRouter functions are not fully supported

241 Page make-up of received pages Same as UTF/UIF

1 Hex dump reception Master trace level – see description below

2 Call control state machine trace Level for general trace

3 Call progress detection trace Reserved

4 Fax Modem trace State changes trace level

242

5 Binary trace (LS1 only) Event trace level

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6 Binary trace (LS1 only) Action trace level

7 Performance counter traces (LS1 only) Reserved

8 Write T.38 diagnostic information to trace00: Default (same as 03)01: No diag info02: Reserved for very compact info03: Write diagnostics of failed calls to tracefile.

9 Write call-control diagnostic information totrace [1]00: Default (same as 03)01: Do not write diagnostic info from call-control.02: Reserved for very compact info.03: Write call control diagnostic info if call-setup fails with an error category other thanREMOTE_BUSY or CALL-NOT-ANSWERED04: Like 03, but include calls that fails witherror category REMOTE_BUSY and CALL-NOT-ANSWERED.

10 Reserved

11 BTR trace level for incoming calls

12 BTR trace level for outgoing calls

243 – 248 Default fax header line Same as UTF/UIF

249 Reference channel Same as UTF/UIF

250 Type of ISDN connection Reserved

251 Bearer and higher layer compatibility offax calls

Flag 0x40 in position 1 is used to set thebearer capability with H.323. All other flagsand positions are ignored.

252 DDI/MSN mode '0': Ignore called party number, accept allcalls.'1': Like MSN with UIF'2': Like DDI with UIF, but overlapped part ofthe called party number will be ignored.

253 Currency units scaling reserved, AOC is not supported

254 – 283 Number conversion table Like UTF/UIF but with some restrictions

284 Call request time-out in seconds Same as UIF

285 DDI length and time-out Reserved, DDI is not supported

286 1-7

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8 Use redirecting number for incoming faxcalls bit 00: Redirecting number disabled (default)1: Redirecting number enabled

Same as UIF

287 -290 ISDN protocol deviations Reserved

1-8 ISDN Layer 1 configuration Reserved

9 Reserved Call peer mask for outgoing calls

10 Reserved Call peer mask for incoming calls

11 Reserved Enable variable Gateway IP

12 Reserved Failover strategy for outgoing calls

13 Reserved Acceptance of inbound calls from unknownIPs

291

14-16 Reserved Reserved


1 Outgoing voice calls00: Disabled (default)01: Enabled

Same as UIF

2 Use configured PABX prefix for outgoingvoice calls:00: No (default)01: Yes

Same as UIF

3 Suppression of redirecting number (donot activate concurrently with generalredirecting number feature, line 286 pos.8 bit 0). Change the most significant bit:0: Redirecting number supported (default)1: Redirecting number suppressed

4 Automatic fax detection00: Disabled (default)else: Time in seconds TCOSS tries todetect an incoming fax after the call hasbeen answered

Same as UIF

5-6 Call transfer modes Reserved. For configuration of Call transfermodes, see the FoIP configuration section inKCS Integration.

295

7-8 Reserved Reserved

296 Log Entry Filtering Same as UTF/UIF


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Description of UFI Specific Configuration LinesConfig line Position Meaning

143 3 Controls which additional info should be written into the author field.

Bits 0 and 1 Used to show transmission parameters, such as “E96” in author field.

00 Never show transmission parameters

01 Show transmission parameters except for failed calls with anextended error info as configured in bits 4-7

02 Show transmission parameters for failed calls with an extendederror info as configured in bits 4-7

03 Show transmission parameters for all calls

Bits 4-7 Defines whether the extended error info (such as I:PROT;…) of a failed callshould be written into the author field (replacing remote CSI/TSI) or not.

00 Recommended default (like 20)

10 Always write the error info in the author field. This mode iscompatible with FoIP releases up to 3.11.02 but is not supportedtogether with the enhanced retransmission mode

20 Generate extended error info for all incoming calls and alloutgoing calls which fails before the 2nd page has beentransmitted.

30 Like 20, but do not generate extended error info if the call failsafter CSI has been received from remote side.

70 Never write the extended error info into the author field.

251 1 Bearer capability for outgoing H.323 calls.

291 9 Call peer mask (default = 00) for outgoing calls

01 First call peer

02 Second call peer

04 Third call peer

08 Fourth call peer

10 Fifth call peer

20 Sixth call peer

40 Seventh call peer

80 Eighth call peer

Any combination Allows to configure which call peers are available for outgoingcalls.

291 10 Call peer mask (default = 00) for incoming calls:

00, FF Incoming calls from all call peers are accepted

01 First call peer

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Config line Position Meaning

02 Second call peer

04 Third call peer

08 Fourth call peer

10 Fifth call peer

20 Sixth call peer

40 Seventh call peer

80 Eighth call peer

Any combination Allows to configure which call peers are used for incoming calls.

Note The incoming call peer is detected according to the IP address used with thefirst message that indicates the incoming call. If an exact match with an IP addressconfigured for a call peer remote address is found, the appropriate call peer numberis used. If no call peer matches, the behavior depends on position 13 of configurationline 291.

291 11 Enable variable gateway IP for outgoing calls

00 Gateway IP is always taken from the Call-peer configuration(default)

01 Variable gateway IP may be specified in the number field.

291 12 Failover strategy for outgoing calls

00 Use current call use the first available call peer (default forupdates) peer until it fails

01

02 Load balance (round robin) between all call peers (default for newchannels)

291 13 Acceptance of inbound calls from unknown IPs

00 Accept inbound calls from IP’s that do not match the IP of anycall-peer (default)

01 Do not accept inbound calls from IP’s that do not match the IP ofany call-peer

Tracing Possibilities in FoIPT.38 specific traces consist of a master trace level and some detailed trace levels. Each detailed tracelevel has its own configuration position. If the value in this position is zero, then the master trace level isused instead of corresponding configuration position. Unless there is any very special situation to trace, itis recommended to activate traces with the master trace level only.

T.38 Traces can be activated with WConfig line editor in the UFI configuration line 242 as describedbelow.

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Position Meaning

1 T.38 Master Trace Level

00 No T.38 trace

01 Less T.38 trace (may be activated on up to 60 channels)

02 Medium T.38 trace (recommended for troubleshooting on some few channels)

03 – or more High T.38 trace (only recommended for intensive troubleshooting on onechannel)

2 Reserved Set to 00

3 Reserved Set to 00

4 State changes trace level

00 Do not trace any state changes in the internal state machine

01 Trace only some major state changes (about 3 lines per document)

02 Trace more state changes

03 Trace all state changes except during data reception/transmission and idle state

04 Trace all state changes except during data reception/transmission

05 or more Trace all state changes

5 Event trace level

00 Trace unexpected events only.

01 Like 00 + events at begin and end of call.

02 Like 01 + events at begin and end of each page

05 or more Trace all events.

6 Action trace level

00 Trace only actions that are called due to unexpected error conditions. (StateTable error handling)

01 Like 00 + one action at begin of transmission or reception

02 Like 01 + actions at begin/end of document/page

03 Like 02 + more actions at begin/end of document/page

04 Like 03 + each received messages for TN

05 Like 04 + actions during idle

06 or more All actions

8 Write T.38 diagnostic information to trace

00 Default (same as value 03)

01 Do not write diagnostic information to trace

02 Reserved for future use

03 Write diagnostic information of failed calls to trace file

9 Write SIP diagnostic information to trace

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Position Meaning

00 Default (same as value 03)

01 Do not write SIP diagnostic information to trace

02 Reserved for future use

03 Write call control diagnostic info if call-setup fails with an error category otherthan REMOTE_BUSY or CALL-NOT-ANSWERED

04 Like 03, but include calls that fail with error category REMOTE_BUSY andCALL-NOT-ANSWERED.

11 BTR trace level for incoming calls. This value is used only if the BtrTraceLevel in FoIP is set to 256(UFI channel specific).

00 Disabled (default)

02 Standard – keep BTR traces for failed calls only

03 Standard – keep BTR traces for all calls.

07 Extended BTR traces (should be used only on request, not recommended forstandard troubleshooting)

12 BTR trace level for outgoing calls. Like Position 11 but for outgoing fax calls.

Binary Trace for Specific OriginatorsYou can activate the Binary Traces for specific originators, as it is possible with Line Server Model 305,by conversion of the received number or caller ID into a number that contains “<T>”. The string “<T>“ isdetected as option to activate the binary trace (with level 3), if it is not already enabled by positions 11 ofconfiguration line 242. In any case, the “<T>“ switch will be removed from the converted number beforethe number is used, for example, for inbound routing.

Note The BTR trace level must be set to 256 (UFI channel specific) in the FoIP configuration to allowactivation of binary traces due to number conversion table.

An example number conversion table is shown below:line 256 '183=83<T> , 257 line 257 '@~41=~41<T> , 258

Line 256 activates the binary trace for all incoming fax calls with DDI/MSN/DID number 83.

Line 257 activates the binary trace for all incoming fax calls where the caller ID contains 41.

Log Entry FilteringLog entries can be filtered according to the type and result of the call. The filter is configured in the configuration line 296, positions 1 to 3:

Position Meaning

1 Sets the filter type for incoming calls

2 Sets the filter type for outgoing calls

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Position Meaning

3 Sets the filter result

Both filters (type and result) must match in order to create a log entry.

For more information, search for “line 296” in the TCOSS System Manual.

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Chapter 5

Migration to KCS FoIP

This chapter describes the basic steps necessary to migrate from other FoIP / VoIP solutions to KofaxCommunication Server Fax over IP (FoIPv3).

Migration from KCS Line Server Model 305The migration of Line Server Model 305 to KCS FoIP (LS1) includes a change of the connection type formanalog (POTS) or ISDN lines to IP. Therefore, it should be handled like a new installation.

The table below provides a feature compare between LS1 and FoIPv3 that excludes the followingfeatures:• Features supported by FoIPv3 only• Features supported by FoIPv3 and LS1• Features specific to the used line type (such as ISDN or VoIP protocols)

Feature LS1 KCS FoIP

Send switch “D” – dial only # Not supported 1

Send switch "t" generate DTMF digits and delaysduring alert state

# Not supported

Send switch "T" generate DTMF digits and delaysduring in connected state

# #2

Use number type, such as TN and TI for outgoing SIP,H.323 calls

# Not supported 3

Test loop for Bit error rate tests # #2

Get Advice of charge (AOC) information # Not Supported

Overlapped reception of Called Party Number (DID orDDI number)

# Not Supported 4

Audio feedback (1 or 3 beeps) for fax server/routingfunctions

# #2

Start fax reception by voice script (SetProperty"faxcall")

# #2

Start fax reception in Voice mode due to CNG # #2, 5

Voice connection priority handling for fail-over voiceservers

# #2

Show operation mode in response field # Not supported

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Feature LS1 KCS FoIP

Store reception speed in received pages # Not supported

Store reception Eye Quality in received pages # Not supported

Set bearer and higher layer compatibility for outgoingcalls

# Partially supported 6

Merge redirecting number into received number with“R”

# Not supported

Note

1. KCS FoIP ignores the switch

2. New features since KCS 10.0

3. The number-type switches are not used by the H.323/SIP signaling but you can convert theseswitches into a prefix number using the outgoing number conversion table.

4. This is typically handled by the software PABX

5. The feature is supported with FoIPv3 only if the used gateway signals the CNG tones using RFC2833 (RTP-NTE) or if the gateway changes to T.38 mode upon detection of CNG.

6. The bearer capability can be set with H.323 calls. The VoIP Gateway is responsible to set thecorrect higher layer and bearer compatibility if the call is forwarded to an ISDN connection.

Migration of FoIPv2 (XCAPI)This section describes the migration of FoIPv2 (XCAPI)

Migrate TCOSS 7.80.04 with XCAPI 3.03.36 (H.323) to FoIPv31. Stop TC/FoIP (XCAPI) via KCS Monitor.

2. Services: Stop XCAPI service and set startup to disabled.

3. Device manager: Disable XCAPI device.

4. Install KCS FoIP.

5. Configure KCS FoIP: Verify that the Reception Mode is set to “TCOSS” and enter a call peer.

6. Start KCS FoIP service via KCS Monitor.

7. Remove TCFOIP from the TCSRV boot section.

Note TCOSS reboot is not required.

Roll Back from FoIPv3 to XCAPI1. Stop KCS FoIP via KCS Monitor.

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2. Remove FOIP from the TCSRV boot section.

3. Device manager: Enable XCAPI device.

4. Services: Enable and start XCAPI service.

5. Insert TCFOIP to the TCSRV boot section.

6. Start TC/FoIP (XCAPI).

Note TCOSS reboot is not required.

Feature Comparison of FoIPv2 and FoIPv3This sections lists the restrictions of the current Kofax Communication Server Fax over IP releasecompared to FoIPv2 and vice versa.

Features Not Supported with FoIPv3Restrictions of the current Kofax Communication Server Fax over IP release compared to FoIPv2.• Allows configuring only one H.323 Gatekeeper

Features Not Supported with FoIPv2Features not supported with FoIPv2 compared to the current Kofax Communication Server Fax over IPrelease.• Fax Polling• Siemens HiPath 8000• Using the send option (Nxx/Exx) to set ECM mode, maximum speed and compression individual for

each call• Voice and Message Wait functionality

Migration of Message Wait via H.323This section describes a sample migration from an existing message waiting solution via TC/MsgWait to anew one based on KCS FoIP.

This migration assumes a working H.323 fax integration with a Cisco CallManager via FoIPv3 asdescribed in Installation of Fax for KCS and Migration of FoIPv2 (XCAPI).

Step 1 – Configure KCS FoIP1. Launch FoIP configuration utility.2. In the “KCS Integration” section, set MessageWait to “MWI control number (Cisco CM, UCM)”.

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Step 2 – Verify New Message Waiting Indication Functionality1. Start the TCfW Communication Server Client2. Send blank messages to the following address:

• free,F:TCMWION;4632;4630

The message waiting indication lamp on the telephone number 4630 will be lit.• free,F:TCMWIOFF;4631;4630

The message waiting indication lamp on the telephone number 4630 will fade.

In the examples given above,• “F:” is a channel group,• “TCMWION” and “TCMWIOFF” are case-sensitive commands,• “4632” and “4631” are the numbers defined in Cisco CallManager for message wait indication and• “4630” is the number of a telephone station.

Step 3 – Change the Prefix of Message Waiting Services

1. In TCfW, from the Admin menu, select Services.

2. Change the MWION service prefix to F:TCMWION;4632;3. Change the MWIOFF service prefix to F:TCMWIOFF;4631;

The user events do not have to be modified.

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Step 4 – Update Arr99 FileIn the Arr99 file, in the ROUTE section, delete or comment out the lines related to old message waitingsolution:**ROUTE,,MWOn:4~,MW:TCMWION;4~;4632,,MWOff:4~,MW:TCMWIOFF;4~;4631

Step 5 – Disable Old Message Waiting1. From the registry key HKEY_LOCAL_MACHINE\SOFTWARE\TOPCALL\Boot\Startup, delete the

line “TCMsgWait”.

2. Stop the TCMsgWait process via KCS Monitor.

Migration of H.323 Voice EngineThis chapter describes a simple use case in which an existing KCS voice installation (native H.323 forCisco CallManager) is upgraded to voice integration via KCS FoIP.

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It assumes a working H.323 fax integration with a Cisco CallManager via FoIPv3 as described inInstallation of Fax for KCS and Migration of FoIPv2 (XCAPI).

Step 1 – Reinstall KCS Server Package1. Before installation, stop the local supervisor service TCSRV via KCS Monitor.

2. Start the KCS Server Package setup. Use release 7.84.00 or later.3. During installation of Voice Server, change the telephone system from “H.323” to “TC Line Server

One”.(When using release 7.87.00 or later, select the “LS1 or FoIP” option.)

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4. Enter the IP address of the server, where FoIP is installed – in our case locally, so either10.18.136.51 or 127.0.0.1. Enter the configured password or leave it empty if no password has beenconfigured.

It is not necessary to reconfigure advanced voice features, MX, LN, and GW integration. Keep theemail reader (TTS) setting unchanged.If you are installing the same Server Package release, it is not necessary to reinstall the voicelib.

Step 2 – Update Registry1. Start Registry Editor and go to My Computer\HKEY_LOCAL_MACHINE\SOFTWARE\Topcall

\TCECP\TCRT\TCE_LS1\LAN000.

2. From My Computer\HKEY_LOCAL_MACHINE\SOFTWARE\Topcall\Boot\Startup, remove the"TCMsgWait" entry.This will be configured in the new VoIP integration.

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Step 3 – Configure KCS FoIP1. Start the KCS FoIP configuration utility. In the KCS Integration tab, select EnabledVoiceServer and

save the configuration.

2. Start KCS Monitor and start the local server (.).

Now you can check if voicemail and voicelink function properly.

HintsIt is not necessary to rerun setup of Voicelink.

Make sure to have outbound voice enabled in UFI channels on KCS for Voice. Change line 295 position 1from “00” to “01”.

Message waiting now uses different addressing (the order of parameters is reversed). Message waitevents / service prefixes have to be adapted (format has changed). For more information, see Step 3 –Create Services and Events in TCfW.

Feature ComparisonThe H.323 Voice feature, "G.729 audio data compression algorithm" is not supported by KCS FoIP.

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Chapter 6

Administration

This chapter summarizes administrative information for troubleshooting Kofax FoIP and lists errormessages and important event log entries.• Errors Codes in TCfW: Displayed in TCfW Communication Server Client• Error Codes in FoIP: Displayed in the internal fax storage• Event Log Entries

Errors Codes in TCfWThis section describes the error codes visible in TCfW Communication Server Client. Two columns areespecially important:• Error: Shows the two-character code of the error.• Response: Shows additional information in case of error.

Two-Character TCOSS Error CodesThe following table gives you an overview about the 2-digit TCOSS error codes that may be produced byUFI. Most error codes are derived from UIF.

Error code Description Notes Normal breakcode

Extended dialingbreak code

IA Call Collision 1 1 1

IC No Channel available (temp.problem)

1 2 1

ID Line disconnect from other stationduring T.30

2, 5 4 1

IF No answer from distant station (ringdetected)

1 2 1

II Error in selection number 1 5 5

IJ No connection to local exchange 1 2 1

IN Wrong number 1 5 5

IS Service or function not implemented 1 5 5

IP Protocol error 1 5 5

IU User Busy 1 2 1

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Error code Description Notes Normal breakcode

Extended dialingbreak code

JO Bad local performance 2, 5 4 1

JP Bad remote performance 3, 4 4 1

JQ Bad Quality of Service (packet loss,delay)

3, 4 4 1

JR Internal Resource error 2 4 1

JS Communication has been stoppedby remote side

3, 4 4 1

JX Unexpected internal error 2 4 1

X0 Call collision in layer 7 1 1 1

X1 Attempt to send an empty document 1 5 5

X2 Error when opening back-receivedfile

1 5 5

X3 Error in back reception 1 5 5

X5 Error during reception 1 4 1

X8 At least one received page has notbeen confirmed

1, 4, 6 n/a n/a

XA Call collision in layer 6 1 1 1

XB Data error within TCI-block 1 5 5

XC Form buffer is out of memory 1 5 5

XF Ring Signal detected 1 2 1

XG Receiver not Ready Time-Out 1, 4 4 4

XK Answerback mismatch 1 5 5

XL No answer from distant telefax 1, 5 5 1

XM ill. identification of called station, 1 4 1

XN Ill. response during training phase 1, 4 4 1

XO Three learn attempts unsuccessful 1, 4 4 1

XP Ill. page confirmation 1, 4 4 1

XR Illegal frame received 1, 4 4 1

XS Unable to find appropriate baud rate 1, 4 4 1

XT No response received 1, 4 4 1

XW Could not find a baud rate duringreception

1, 4, 6 n/a n/a

XX Illegal Identification of Tx Station 1, 4, 6 n/a n/a

XY Command expected but notreceived during reception

1, 4, 6 n/a n/a

XZ Illegal command received 1, 4, 6 n/a n/a

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Note

1. Same as with UIF

2. Error code was already defined for FoIPv2

3. New error code

4. This error code is not available with FoIPv2

5. This error code is less specific with FoIPv2

6. Error code is used for reception only

Additional Error DetailsThe Error Codes lists and describes the error codes that might occur during fax transmission. The errorcodes displayed in the FoIP web portal are also available in TCfW in the Response column. This field haslimited size (max. 24 characters), therefore, the information is abbreviated.

For a successfully delivered message, the Response field displays the CSI number (Called StationIdentification; for outgoing messages) or the TSI number (Transmitting Station Identification; for incomingmessages).

In case of an error, the Response field provides information about the error in the following syntax:

I:<category (short name)>;<connection level>;<error code>;<diagnostics>

Sample error messages:I:HUMAN;55;21321;MCI:TARGET;20;12404;8-205

Error Codes in FoIPThe result of fax communication is described using the following four fields.• Category: Fixed list of simple error reasons• Code: More detailed error code. It may be mapped via language specific to a user friendly text. The

error code should not be used for application program decisions. These programs should use the errorcategory and/or connection level.

• Connection Level: This is a value between 0 and 100 that defines the maximum connection level thatcould be reached within the call.

• Diagnostics: Additional free text that provides more information about the error. It may be used fortroubleshooting. It should only be presented to technical end-users.

Error CategoriesThe following error categories are defined:

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Name (short name) Description Examples

OK (OK) The document has been deliveredsuccessfully.

BAD-CONTENT (CONTENT) The document which should be sentcontains a bad or not supportedcontent

Invalid contentNot supported image format

BAD_LOCAL_PERF (LPERF) Communication failed because localsystem was not able to meet real-time constraints.

Insufficient CPU performance

BAD_REMOTE_PERF (RPERF) Communication failed becauseremote system was not able to meetreal-time constraints

Bad timing generated by the remoteside

RESTRICTION (RESTR) The requested operation isnot supported with the currentconfiguration, license or usercredentials.

Missing licenseBad user ID / password

LOCAL-BUSY (LBSY) No local channel is temporaryavailable.

No local channel is available.

NETWORK-BUSY (NBSY) Parts of the network are temporarynot available

GW/PBX has no line to localexchange

REMOTE_BUSY (RBSY) Remote station is busy Busy from distant user

CALL-NOT-ANSWERED (NOANS) The call has not been answered(alerting time-out)

Distant user has been alerted but hasnot answered the call

NETWORK-UNAVAILABLE (NONET) Connection to network fails Failed to connect to FoIP GW No lineto exchange

BAD-TARGET (TARGET) The number is wrong Invalid characters in the number fieldNot assigned number

INCOMPATIBLE-TARGET (INCOMP) The target exists but it is notcompatible

Incompatible DIS has been received

HUMAN-TARGET (HUMAN) A human has been reached insteadof fax machine

Call has been answered but no faxhas been detected (XL)

WRONG-TARGET (WRONG) The target is compatible but not theexpected.

CSI verification failed

SERVER-POWER-FAILURE(PWROFF)

Communication failed due to powerfailure, restart or server crash.

Power failure/Restart or crash duringcommunication

CANCELED-LOCAL (CANC-L) Operation has been canceled bylocal user/application

Shutdown during communicationOperation has been canceled by user

CANCELED-REMOTE (CANC-R) Operation has been canceled byremote user/application

Remote user has stopped the faxmachine.

EXPIRED (EXPIRED) Operation has been canceled byscheduler

RESOURCE (RES) The requested operation could notbe performed due to (temporary)resource restrictions

Out of memoryDisk full

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Name (short name) Description Examples

BAD-LINE- QUALITY (BLINE) Communication failed due to bad linequality

RTN received

BAD-IP-QOS (BNET) Quality of Service (Packet loss,delay) between FoIP and gatewaywas insufficient

High packet loss

PROTOCOL (PROT) Communication of media data (usingprocedure, such as T.30) failedCall setup failed with an error

Most T.30 error reasonsTime-out during call establishment

INTERNAL (INT) Any other internal or unexpectederror

UNKNOWN (UNKN) Category is not specified.

Error CodesThe following table gives you an overview about the 2-digit TCOSS error codes that may be produced byUFI. Most error codes are derived from UIF.

Code Description

10000 unspecified error in Call Control TNC

11000 unspecified SIP error code = Offset for Q.850 codes

11001 unallocated (unassigned) number

11002 no route to specified transit network

11003 no route to destination

11004 send special information tone

11005 misdialed trunk prefix

11006 channel unacceptable

11007 call awarded and being delivered in an established channel

11008 preemption

11009 preemption - circuit reserved for reuse

11016 normal call clearing

11017 user busy

11018 no user responding

11019 no answer from user (user alerted)

11020 subscriber absent

11021 call rejected

11022 number changed

11023 redirection to new destination

11025 exchange routing error

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Code Description

11026 non-selected user clearing

11027 destination out of order

11028 invalid number format (address incomplete)

11029 facility rejected

11030 response to STATUS ENQUIRY

11031 normal, unspecified

11034 no circuit/channel available

11035 call queued

11038 network out of order

11039 permanent frame mode connection out of order

11040 permanent frame mode connection operational

11041 temporary failure

11042 switching equipment congestion

11043 access information discarded

11044 requested circuit/channel not available

11046 precedence call blocked

11047 resource unavailable, unspecified

11049 quality of service not available

11050 requested facility not subscribed

11052 outgoing calls barred

11053 outgoing calls barred within CUG

11054 incoming calls barred

11055 incoming calls barred within CUG

11057 bearer capability not authorized

11058 bearer capability not presently available

11062 inconsistency in designated outgoing access information and subscriber class

11063 service or option not available, unspecified

11065 bearer capability not implemented

11066 channel type not implemented

11069 requested facility not implemented

11070 only restricted digital information bearer capability available

11079 service or option not implemented, unspecified

11081 invalid call reference value

11082 identified channel does not exist

110


Code Description

11083 a suspended call exists, but this call identity does not

11084 call identity in use

11085 no call suspended

11086 call having the requested call identity has been cleared

11087 user not member of CUG

11088 incompatible destination

11090 non-existent CUG

11091 invalid transit network selection

11095 invalid message, unspecified

11096 mandatory information element is missing

11097 message type non-existent or not implemented

11098 message not compatible with call state or message type non-existent or not implemented

11099 information element/parameter non-existent or not implemented

11100 invalid information element contents

11101 message not compatible with call state

11102 recovery on timer expiry

11103 parameter non-existent or not implemented, passed on

11110 message with unrecognized parameter, discarded

11111 protocol error, unspecified

11127 interworking, unspecified

11200 none Q.859/H.225/H.245 error

11201 unspecified SIP error code = Offset for Q.850 codes

11200 error codes from OpenH323 library = Offset for OpenH323 codes

11201 local endpoint application cleared call

11202 local endpoint did not accept call OnIncomingCall()=FALSE

11203 local endpoint declined to answer call

11204 remote endpoint application cleared call

11205 remote endpoint refused call

11206 remote endpoint did not answer in required time

11207 remote endpoint stopped calling

11208 transport error cleared call

11209 transport connection failed to establish call

11210 gatekeeper has cleared call

11211 call failed as could not find user (in GK)

111


Code Description

11212 call failed as could not get enough bandwidth

11213 could not find common capabilities

11214 call was forwarded using FACILITY message

11215 call failed a security check and was ended

11216 local endpoint busy

11217 local endpoint congested

11218 remote endpoint busy

11219 remote endpoint congested

11220 could not reach the remote party

11221 the remote party is not running an endpoint

11222 the remote party host off line

11223 the remote failed temporarily app may retry

11224 the remote ended the call with unmapped Q.931 cause code

11225 call cleared due to an enforced duration limit

11226 call cleared due to invalid conference ID

11227 call cleared as OSP server unable or unwilling to route

11228 call cleared as number was invalid format

11229 call cleared due to unspecified protocol error

11300 unspecified H.225 error

11400 unspecified H.245 error

11401 Remote party does not accept T.38

11500 errors from TNC_H323 code (our error codes)

11501 no answer to outgoing SETUP within time

11502 no gateway IP specified/configured

11503 maximum number of configured calls active

11504 called party number is missing

11505 called party number is invalid

11506 outgoing call has been disconnected (with normal call cleared) before Alert

11507 outgoing call has been disconnected (with normal call cleared) after Alert but without connect

11720 Unspecified problem during control-call

11721 Control call function is not supported

12000 unspecified SIP error code = Offset for SIP codes

12300 multiple choices

12301 moved permanently

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Code Description

12302 moved temporarily

12305 use proxy

12380 alternative service

12400 bad request

12401 unauthorized

12402 payment required

12403 forbidden

12404 not found

12405 method not allowed

12406 not acceptable

12407 proxy authentication required

12408 request timeout

12410 gone

12413 request entity too large

12414 request-URI too large

12415 unsupported media type

12416 unsupported URI scheme

12420 bad extension

12421 extension required

12423 interval too brief

12480 temporarily not available (phone was ringing, but nobody picked up the call)

12481 call leg/transaction does not exist

12482 loop detected

12483 too many hops

12484 address incomplete

12485 ambiguous

12486 busy here

12487 request terminated

12488 not acceptable here

12491 request pending

12493 undecipherable

12500 internal server error (also observed when distant phone was busy)

12501 not implemented

12502 bad gateway

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Code Description

12503 service unavailable

12504 server time-out

12505 SIP version not supported

12513 message too large

12600 busy everywhere

12603 decline

12604 does not exist anywhere

12606 not acceptable

12700 transaction layer timeout (proprietary extension, not defined in RFC 3261)

12701 Incoming INVITE transaction failed because ACK was not received within time-out

12720 Unspecified problem during control-call

12721 Control call function is not supported

12722 Inbound call to Multicast IP has been taken be other instance

12731 sips URL security was downgraded by remote side to sip

20000 unspecified error in a media Control TNC

21000 unspecified error in a T.38 TNC

21100 unspecified error during T.30 reception

21102 line has been disconnected during reception before T.38 mode

21103 failed to switch to T.38 mode within time-out

21104 Gateway did not changed mode to T.38 during reception (replaced byT30_RX_TIMEOUT_NO_CNG since Pass-Through support)

21105 FoIP reception was started in G.711 pass-through mode but no G.711 RTP packets werereceived

21106 FoIP reception was started in T.38 mode, but no T.38 packets were received

21110 unexpected time-out during reception

21111 no DCS received within timeout

21112 no DCS within timeout after reception of CNG

21113 sender did not send training retry after FTT response

21114 sender did not send page data after CFR response

21120 unexpected line disconnect during reception

21121 line has been disconnected by other side before CNG

21122 line has been disconnected by other side between CNG and DCS

21123 line has been disconnected by other side between DCS and CFR

21124 line has been disconnected by other side between DIS and first page

21125 line has been disconnected by other side between MPS/MCF and next page

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Code Description

21130 unexpected T.30 DCN during reception

21131 T.30 DCN has been received from sender instead of DCS

21132 T.30 DCN after MPS instead of fax page received

21141 at least one received page has been rejected and at least on page has been confirmed

21142 non-ecm-data received instead of end-of-page command

21161 at least one received page has been rejected and at least one page has been confirmed

21162 all received pages have been rejected

21163 bad pages received in ECM mode

21164 Remote fax aborted none-ecm transmission after none-final page due to bad line conditions

21165 Sender stopped ecm transmission (due to bad line conditions) with EOR or DCN

21170 call failed due to bad local performance during reception

21171 call failed due to bad local performance during storage of non-ECM reception page

21300 (1300) unspecified error during T.30 transmission

21301 the received CSI does not match with the expected CSI

21302 line has been disconnected during transmission before T.38 mode

21303 failed to switch to T.38 mode within time-out

21304 Gateway did not changed mode to T.38 during transmission

21305 FoIP transmission was started in G.711 pass-through mode but no G.711 RTP packets werereceived

21306 FoIP transmission was started in T.38 mode, but no T.38 packets were received.

21310 unexpected time-out during transmission

21311 no DIS/CED received within timeout

21312 CED received but no DIS within timeout

21313 no response has been received to DCS/TCF

21314 no page response received after 3 MPS/EOP/EOM (end of none-ECM page) tries. MCF/RTP/RTN response was expected

21315 no page response received after 3 PPS (partial page signal) tries. RNR/MCF response wasexpected

21316 no page response received after 3 RR (receiver ready) tries. RNR/MCF response was expected

21317 no page response received after 3 CTC (continue to correct) tries. CTR response was expected

21320 line has been disconnected (unspecified)

21321 line has been disconnected w/o DIS/CED reception

21322 line has been disconnected w/o DIS but after CED

21323 line has been disconnected during training

21324 line has been disconnected after high-speed turn-on but before transmission of data

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Code Description

21330 T.30 DCN has been received (unspecified)

21331 T.30 DCN has been received (CFR expected)

21332 none-ecm page has been rejected with DCN message

21333 ecm page has been rejected with DCN message

21334 T.30 DCN has been received without DIS

21341 transmission failed because the receiver is compatible with the selected mode

21351 command could not be sent because no silence has been detected within time-out

21352 DCS could not be send because no silence has been detected within time-out

21361 failed to find a compatible speed during training

21362 page has been rejected with RTN

21364 ecm-page has been rejected with RTN. This may be caused by coding problems

21365 page retransmissions (with ECM) have been stopped because line quality is too bad

21366 none-ecm page accepted with RTP but speed reduction for next page was not possible or failed

21380 call failed due to bad remote performance of receiving fax machine

21381 call failed because receiving fax machine failed to confirm reception of non-ECM page withintimeout

31000 unspecified SMTP outbound error = Offset for SMTP error codes (1xx - 5xx)

40000 unspecified error in any Communication WF TNC

41000 unspecified error in FaxMain TNC

42000 unspecified error in Fx7 TNC

42001 no enabled call peer found

42010 Any internal component disconnect unexpected

42011 connection to call control component has been lost

42012 connection to call control component has been lost

42020 the component has received an unexpected internal event

42101 incoming call failed because it has not been disconnected before it could be accepted by anyfree channel

42102 reception failed because no image line has been received within 18 minutes

42103 Incoming call has been rejected because there is no connection to TCOSS

42151 incoming voice call failed because component for Voice-Server integration failed or was disabled

42152 failed to create reception documentation

42201 Connect was received with control-call. This was not expected and may be caused be wrongcalled number

42410 Bad inbound number unspecified

42411 Bad syntax of inbound number (such as invalid command)

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Code Description

42412 The specified inbound address does not exist

42413 The specified user or password is not valid

42414 A 7x command failed

42415 the requested function is not allowed with current security configuration

43000 unspecified error in Fx7 TNC

43103 Incoming call has been rejected because no Voice Server is connected

44000 unspecified error in TNC_Albin

44010 root XML not well formed

44020 Message conversion exit failed

44030 Unexpected (bad) message content received (such as empty TIFF from RightFax Server)



60001 folder export failed

Connection LevelThe following table describes the defined connection levels.

Value Description Typical error examples

0 Connection level is unknown Default if no level is used

1 Connection to some internal components ismissing

Connection to H.323 component failed

10 Local resources, such as line or socket, to initiatea call have been allocated. A communicationmedia, such as physical line, with the network hasbeen selected

No connection the GatewayISDN Establish Layer 1

15 Any low level, such as ISDN Layer 1, connectionto a network could be established.

TCP/IP Connection (or Layer 1/2 Connection) toGateway/PBX could be established

20 Communication with network call control layer waspossible

SETUP_ACK, ALERT, CONNECT, .. message hasbeen received

25 Network connection is busy Busy from network

30 Any signal that may indicate that the remote sidemay be available has been received. It indicatesthat the number is valid.

Busy from remote side

40 The remote side has been alerted

45 Call has been rejected by remote side

50 Call control connection has been established

55 Media channel could be opened

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Value Description Typical error examples

60 A compatible media channel protocol has beendetected.

Error after CED/CNG or V.21 Flags have beendetected.

65 Media channel protocol negotiation has beenstarted

Error after DIS or DCS command has beenreceived from remote side.

70 Media channel protocol, such as T.30 connectioncould be established.

80 Data has been sent / received. It may have beenreceived from the other side

90 At least one page has been confirmed

100 The complete activity has been successfullycompleted.

Event Log EntriesWhen Kofax Communication Server Fax over IP and TCOSS establish a connection, both TCOSS andFoIP generate event log entries. This chapter describes the entries created by TCOSS that speak aboutthe status of the connection.

The entries are written to the Application event log. To display the log, click Start | Run, type “eventvwr”and click OK; then select Application.

When TCOSS successfully connects with FoIP, the following event log entry is written to the log file:

Info: TCOSS has been connected to FoIPv3 1.08.16 in SSLV(L.10)-SLVSTART

This information confirms that the connection has been established and TCOSS is ready to send andreceive faxes via FoIP.

When FoIP is erroneously configured to receive faxes into local storage instead of TCOSS, the followingerror is written to the log:

A channel stopped working due to wrong configuration, insufficient license orline errors. Error message: TCOSS has been connected to FoIPv3 1.08.16 usingreception to local storage! Change FoIPv3 configuration to enable reception toTCOS in SSLV(L.10)-SLVSTART Refer to TCOSS system manual for more details.

In this case, FoIP configuration must be changed. See the General section of Basic Configuration.

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Chapter 7

Development Tools

In addition to the functionality described in chapter Web Portal, FoIP offers a set of development tools.Connect to the address http://localhost:25081/file/index2.html and the following items areadded to the menu on the left side.

Send Fax in Real TimeClick the Send in Real Time link from the menu to display the following screen.

The following table provides a more detailed description of the parameters.

Value Description

Number to Dial Destination fax number is the only required parameter

Gateway Host Destination gateway with an optional prefix "h323:" or "sip:" (H.323 is used if no prefix isspecified; configured H.323 gateway is used if you leave this empty)

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Value Description

Caller ID Calling party number

Expected CSI Expected called subscriber identification, or a part of it, for verification that the correctdestination has been called

Source TSI Originating subscriber identification

Sender Name Name of the sender displayed in the fax header line

TIFF File TIFF file to be sent (a built-in default file is sent if you leave this field empty). See SupportedTIFF Options for format details.

To send a test fax, enter the destination fax number (and possibly the other optional parameters), choosea TIFF file that should be faxed (or leave the field blank to use a default file), and send the messageby clicking the “Send Fax” button. You might use the “Preview” button before sending to view the faxmessage including the generated header line.

A message delivered in real time is not added to the fax storage. Instead, message delivery startsimmediately and the browser window waits for a response from the server.

After a few minutes (1 page of fax requires approximately 40 seconds to deliver), the following message isdisplayed in the bottom frame:

T.38 PlayerNote The information from this chapter and the functionality described therein is important for a short listof use cases and therefore is only marginally useful for most users of the FoIP.

The player allows to replay recorded T.38 fax sessions. Having a Wireshark network trace with a coupleof T.38 fax calls recorded on (for example) a customer site, it is now possible to reproduce specific faxtransmission problems that have occurred on that customer site.

T.38 player supports two input formats for the fax session:• PCAP: This is the direct result produced by Wireshark. It is a binary file. To replay a PCAP file, it must

contain only the send portion (T.38 data sent from the fax originator) of a recorded Wireshark T.38fax session to another fax recipient. It may contain only the transmitted (Tx) T.38 UDP packets. Thisfiltering must be performed when saving the trace in Wireshark.

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• T.38 XML: This format can be created from the binary PCAP files by the T.38 PCAP Converter. Theconverter can filter either transmitted (Tx) or received (Rx) packets and both can be replayed using theplayer.

Prepare the T.38 Fax Stream to ReplayTo prepare a compatible T.38 fax stream, you have to extract the UDP (T.38) data stream sent from thefax originator to the recipient of a particular fax call recorded in the Wireshark trace.

Note If you intend to use the T.38 PCAP Converter to generate XML from the recorded traces, you cansave both directions of the fax communication.

1. Find the particular T.38 fax call in the Wireshark trace. Refer to the “How to Find a Call in theWireshark Traces” section in TC/FoIP Tracing Technical Manual.

2. Once you have identified your call by the means of the call graph, inspect the “TSI” line of that call(the first T.30/T.38 packet sent from the fax originator), and note down its source and destinationports. In the example below, the source port is 17796 and the destination port is 1750.

3. Write the following filter into the display filter line and click Apply:

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4. Click File | Save as to save the extracted T.38 data into a separate file. Make sure that you selectDisplayed under Packet Range.

Initiating a Call1. To replay the fax session, click the T.38 Player link from the menu. The following page is displayed.

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2. Configure the parameters of this function:

Telephone Number The number that is dialed over the configured gateway (see Gateway IP in BasicConfiguration). This parameter must be numeric and may not be longer than 70 digits.

Initial Delay in ms The delay after which the first T.38 packet is supposed to be sent. This may help tosynchronize the fax send simulator with the receiving (real) fax machine

Check ReceivedFrames

If selected, checks received frames only. This option is only available for XML input.

Enter the Trace File The path to the extracted Wireshark T.38 data stream (as described above).

Note A full Wireshark without any extractions does not work properly. Alternatively,you can enter a T.38 XML file.

Note Enter at least a telephone number and the file name.

3. Click Send Fax to start the fax transmission. A response is displayed in the lower window, whichconsists of an informational table showing call ID, state and disconnect reason, and Refresh andDisconnect buttons.

Showing Current State of the CallWhen you click Refresh, the information in the table is updated. If the call gets alerted, the state columnchanges as shown below.

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When clicking Refresh, after the call was disconnected, the disconnect reason is added in the table. If,for example, the called device has ended the call without any error, the Disconnect info string indicates“unknown”.

Interrupting a CallBy clicking Disconnect at any time during an active call, the call is disconnected immediately. You canverify this by clicking Refresh and thus updating the call information.

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Fax Replay LimitationsAs the Fax Replay function simply replays recorded fax data from the trace file (and ignores the fax datacoming from the fax recipient), a successful fax transmission cannot be guaranteed.

If, for example, the fax recipient asked for any retransmission for any reason or even requested a differentfax transmission mode (like non-ECM instead of ECM), the fax replay would fail.

Outgoing calls are established using the H.323 protocol. SIP is not supported.

T.38 PCAP ConverterThe T.38 PCAP Converter tool can be used to convert a PCAP file to a T.38 XML file.

PCAP files are binary files, and are complicated to read and edit manually. On the other hand, theXML format displays the same information in a user-friendly way. The converter also allows to filter theconversation into sent and received packets.

The converter runs only locally.

1. Configure the parameters of this function.

Enter Target File The name of the destination XML file.

IP Address The IP address of the one party of the fax session that you want to focuson.

Ports The port numbers for transmitting (TX) and receiving (RX). By enteringonly one of these ports, you can limit your XML file to either transmitteddata or received data.

Ignore Redundant Frames If selected, omits repeating frames from the fax session.

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Relative Timestamps By default, the time stamps assigned to frames show absolute time, that is,how much time occurred since the first frame.If this option is selected, the time stamps assigned to frames switches torelative time stamps, that is, how much time occurred since the previousframe. (This allows simple changes in the order of the frames.)

Enter Source File The name of the source PCAP file. This file may only contain T.38 UDPpackets.

2. Click Start.

Test Server1. Click the Test Server link in the menu. The following page is displayed.

2. From the Template list, select the template you need. All other fields on this page including thepredefined function calls to FoIP are updated according to the selected template. The generatedURL that represents the function call is also indicated.

3. If Open response in a new browser window is selected, the result is written into a new browserwindow. Otherwise, the result is written into the frame at the bottom of the page.

4. Click one of the following:• Show Formatted Result: Converts the result of the function call using a style-sheet or content

type if one of them is available.• Show XML Result: Always returns the response in XML format, which proves very useful when

you want to see the complete response for each function.

Note This function exists only for debugging purposes. It is not intended for the normal user and istherefore described rather superficially.

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Chapter 8

Web Service Functions

Kofax Communication Server Fax over IP uses a web service interface for communication with faxservers. The functions of this interface are defined in the “fax.wsdl” file. The functions are described indetail in a separate document called Web Services for Fax.

The web service interface is generic and is used in several connectors. This chapter lists only exceptionsspecific to FoIP.• In the SendFax function, the parameter Path can be used to refer to a gateway. The parameters

CodePage and ServiceName are ignored.• In the GetFax and GetFax20 functions, the element Filter and its sub-elements are ignored.

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Chapter 9

Specifications

This chapter describes the TIFF specifications supported with FoIP.

Supported TIFF OptionsThe TIFF 6.0 specification allows a large number of image options. This section defines the subset ofTIFF images which can be sent as fax documents.

Only the “bilevel” (black and white) TIFF image type is supported. Additionally, the TIFF tags listed in thefollowing table are restricted to the specified values:

TIFF tag Supported values

Compression (259) 1: No compression2: Modified Huffman run length encoding3: Group 3 Fax (MH, MR encoding)4: Group 4 Fax (MMR encoding)32773: PackBits compression

ImageWidth (256) 1728 or less (pixels per scanline)Images smaller than 1728 pixels are not scaled but padded with additional left and rightmargins to reach the 1728 pixel width required for transmission

XResolution (282) 50 to 300 (dpi)

YResolution (283) 50 to 300 (dpi)

The TIFF file may contain one or more images which correspond to individual fax pages. Both little-endianand big-endian byte order of the TIFF file is supported.

How to Create a TIFF File for FaxingYou can use a plethora of tools to generate a TIFF file as specified in Supported TIFF Options. Thefollowing tools were tested:• Microsoft Office Document Image Writer• KCS Printer Driver• KCS Image Conversion Library – TCIMG32

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Microsoft Office Document Image WriterThe Document Image Writer is a printer driver included in Microsoft Office 2003 and later versions. Youcan print any Office document to this printer to create a valid TIFF file suitable for faxing via FoIP.

1. In the Print dialog, select Microsoft Office Document Image Writer from the list of availableprinters

2. Click Properties.

3. Click the Advanced tab.

4. In the Output format, select TIFF – Monochrome Fax with the resolution Fine (200 DPI).

5. Confirm the configuration by clicking OK and then proceed with printing as usually. A new TIFF filewill be created.

KCS Printer DriverThe KCS Printer Driver for Windows 2000/XP/2003/Vista offers the following TIFF formats compatible withFoIP:• TIF0 – Packbit (default)

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• TIF1 – Uncompressed• TIF2 – CCITT G3 Type 2 / 1D-Code without EOL• TIF3 – CCITT G3 Type 3 / 1D-Code with EOL• TIF4 – CCITT G4 Type 4 / 2D-Code without EOL

TCIMG32The KCS image conversion library TCIMG32 is a part of TC/Link and TCfW installations. The followingTIFF outputs can be used for faxing via FoIP.

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Chapter 10

Acknowledgments

This product includes software developed by the OpenH323 project. (http://www.openh323.org, http://www.voxgratia.org).

The product also makes use of the 7-zip archiver (http://www.7-zip.org/). See the C:\Program Files\Kofax\Fax Connector\FoIP\00\acknowledgments subdirectory for license information. Theversion used in FoIP can be downloaded here: http://www.topcall.com/sources/7za442.zip.

Uses OpenSSL for secure communication (http://www.openssl.org/).

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http://www.openh323.org

http://www.voxgratia.org

http://www.voxgratia.org

http://www.7-zip.org/

http://www.topcall.com/sources/7za442.zip

http://www.openssl.org/

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