Professional Documents
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Date: 2019-12-13
© 2019 Kofax. All rights reserved.
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transmitted in any form without the prior written permission of Kofax.
Table of Contents
Preface...........................................................................................................................................................7
Related Documentation................................................................................................................................. 8
Chapter 1: Introduction.............................................................................................................................. 9
Voice and Fax over IP....................................................................................................................... 9
Kofax Solution.....................................................................................................................................9
Use Cases........................................................................................................................................ 10
Installation with Kofax Communication Server......................................................................10
Standalone Configuration...................................................................................................... 11
International Standards and Recommendations.............................................................................. 11
IP Infrastructure Requirements..............................................................................................11
Compatibility...........................................................................................................................12
Unicode Support.................................................................................................................... 12
Chapter 2: Installation.............................................................................................................................. 13
Prerequisites..................................................................................................................................... 13
Setup.................................................................................................................................................15
Personal Firewall Settings..................................................................................................... 18
Standalone Installer (FoIP.msi)........................................................................................................ 19
Uninstalling KCS FoIP........................................................................................................... 20
Reboot Warning When Upgrading FoIP on Windows Server 2008...................................... 20
Upgrade from Older Releases......................................................................................................... 21
Upgrade from KCS FoIP 3.11.xx – 3.18.xx (KCS 9.0 – 9.2).................................................21
Upgrade from KCS FoIP 3.06.xx (KCS 8.2)......................................................................... 21
Basic Configuration...........................................................................................................................21
General...................................................................................................................................22
List of Call Peers................................................................................................................... 22
Advanced Configuration................................................................................................................... 23
VoIP/FoIP Network.................................................................................................................23
Voice.......................................................................................................................................26
Fax......................................................................................................................................... 27
H.323 Signaling......................................................................................................................28
SIP Signaling......................................................................................................................... 29
SSL Certificate....................................................................................................................... 31
HTTP...................................................................................................................................... 32
KCS Integration......................................................................................................................33
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Storage Settings.................................................................................................................... 34
Advanced............................................................................................................................... 34
Saving Configuration........................................................................................................................ 36
Running FoIP in a Console Mode................................................................................................... 37
Policy-based Quality of Service....................................................................................................... 38
Chapter 3: Web Portal.............................................................................................................................. 44
Call Peer Statistics........................................................................................................................... 45
Send Fax via Storage...................................................................................................................... 46
Sending in Loop-Back Mode................................................................................................. 47
Fax Header Line.................................................................................................................... 48
Fax Storage...................................................................................................................................... 49
Maintaining Fax Storage........................................................................................................49
Inbound Messages.................................................................................................................50
Outbound Messages..............................................................................................................51
Chapter 4: Integration with KCS............................................................................................................. 53
Installation of Fax for KCS...............................................................................................................53
Step 1: Set Up TCOSS Server for FoIP............................................................................... 53
Step 2: Set Up FoIP..............................................................................................................54
Step 3: Verify Installation.......................................................................................................54
Step 4: Set Up Call Peers.................................................................................................... 55
Installation of Message Wait for KCS.............................................................................................. 55
Step 1: Configure KCS FoIP................................................................................................. 56
Step 2: Configure CallManager and Test Message Waiting................................................. 56
Step 3: Create Services and Events in TCfW...................................................................... 56
Message Wait via SIP Notify (RFC 3842).............................................................................57
Installation of Voice for KCS............................................................................................................ 57
Install Voice Server................................................................................................................57
Configure Voice Support in FoIP...........................................................................................59
Enable Voice in the UFI Channels........................................................................................ 59
Test the First Incoming Voice Call........................................................................................ 60
Leave a Messages via Telephone.........................................................................................60
Test Outgoing Voice Call with TC/Player.............................................................................. 60
Unique Single Number Support............................................................................................ 61
Hints....................................................................................................................................... 62
Supplementary Services...................................................................................................................62
Call Diversion (Redirecting Number).....................................................................................62
DTMF Reception....................................................................................................................64
DTMF Generator for Two-Stage Outbound Dialing...............................................................64
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Call Transfer.......................................................................................................................... 65
Configuration Hints........................................................................................................................... 65
Setting Own Calling Party Number....................................................................................... 65
Fax on Demand..................................................................................................................... 66
High Availability Support........................................................................................................66
SIP Registrar and H.323 Gatekeeper................................................................................... 73
Behavior If an Incoming Call Is Rejected..............................................................................74
FoIP and Internet Telephony Service Providers....................................................................74
V.34 Fax.................................................................................................................................75
Support Variable Gateway IP for Outgoing Calls.................................................................. 76
Support of Static NAT............................................................................................................77
Endpoint Name...................................................................................................................... 78
T.38 vs. G.711 Pass Through Fax........................................................................................ 78
T.38 Redundancy................................................................................................................... 79
Encryption of VoIP/FoIP Network Traffic............................................................................... 80
Voice Prompts........................................................................................................................87
Multiple FoIP Instances......................................................................................................... 88
Configuration Values.........................................................................................................................88
UFI Channel Configuration.................................................................................................... 88
Description of UFI Specific Configuration Lines....................................................................92
Tracing Possibilities in FoIP.................................................................................................. 93
Log Entry Filtering................................................................................................................. 95
Chapter 5: Migration to KCS FoIP...........................................................................................................97
Migration from KCS Line Server Model 305....................................................................................97
Migration of FoIPv2 (XCAPI)............................................................................................................98
Migrate TCOSS 7.80.04 with XCAPI 3.03.36 (H.323) to FoIPv3.......................................... 98
Roll Back from FoIPv3 to XCAPI.......................................................................................... 98
Feature Comparison of FoIPv2 and FoIPv3......................................................................... 99
Migration of Message Wait via H.323..............................................................................................99
Step 1 – Configure KCS FoIP...............................................................................................99
Step 2 – Verify New Message Waiting Indication Functionality.......................................... 100
Step 3 – Change the Prefix of Message Waiting Services................................................. 100
Step 4 – Update Arr99 File................................................................................................. 101
Step 5 – Disable Old Message Waiting.............................................................................. 101
Migration of H.323 Voice Engine................................................................................................... 101
Step 1 – Reinstall KCS Server Package............................................................................ 102
Step 2 – Update Registry....................................................................................................103
Step 3 – Configure KCS FoIP.............................................................................................104
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Hints..................................................................................................................................... 104
Feature Comparison............................................................................................................ 104
Chapter 6: Administration...................................................................................................................... 105
Errors Codes in TCfW....................................................................................................................105
Two-Character TCOSS Error Codes................................................................................... 105
Additional Error Details........................................................................................................107
Error Codes in FoIP....................................................................................................................... 107
Error Categories...................................................................................................................107
Error Codes..........................................................................................................................109
Connection Level................................................................................................................. 117
Event Log Entries........................................................................................................................... 118
Chapter 7: Development Tools.............................................................................................................. 119
Send Fax in Real Time.................................................................................................................. 119
T.38 Player......................................................................................................................................120
Prepare the T.38 Fax Stream to Replay............................................................................. 121
Initiating a Call..................................................................................................................... 122
Showing Current State of the Call...................................................................................... 123
Interrupting a Call................................................................................................................ 124
Fax Replay Limitations........................................................................................................ 125
T.38 PCAP Converter..................................................................................................................... 125
Test Server......................................................................................................................................126
Chapter 8: Web Service Functions....................................................................................................... 127
Chapter 9: Specifications....................................................................................................................... 128
Supported TIFF Options.................................................................................................................128
How to Create a TIFF File for Faxing............................................................................................128
Microsoft Office Document Image Writer............................................................................ 129
KCS Printer Driver............................................................................................................... 129
TCIMG32..............................................................................................................................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 and
VoIP in their environment.
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Related Documentation
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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Chapter 1
Introduction
This chapter describes fax and voice over IP solutions in general, the architecture and use cases of Kofax
Communication Server Fax over IP and international standards and recommendation.
In 1998 ITU released the T.38 recommendation dealing with the transmission of faxes over IP networks
in real time. This standard guarantees interoperability between fax over IP terminals. T.38-compatible
gateways are used to adopt the T.38 protocol to PSTN (typically via ISDN) so that communication with
conventional Group 3 fax machines in the public PSTN is possible. Today, T.38 is an industry standard
which 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. This
kind of operation does not require T.38 support by the IP environment but it is much more sensitive for
network jitter, packet loss and clock synchronization problems.
Kofax Solution
The fax over IP solution communicate with the H.323 or SIP network. It supports FoIP using T.38
recommendation, FoIP G.711 pass through mode and VoIP.
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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 next
section on Use Cases). Incoming voice calls are supported via TCOSS connection only.
Use Cases
Kofax Communication Server Fax over IP was designed for two main scenarios:
• Installation with Kofax Communication Server
• Standalone configuration
The scenario with TCOSS involves at least three of the five FoIP components: TCOSS connection, FoIP
core, and connection to call peers, such as a FoIP gateway. Outgoing messages from TCOSS are sent
in real time (without storage) to their intended recipient. Similarly, incoming messages are delivered to
TCOSS.
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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 solution
and for troubleshooting.
FoIP can also be configured to transfer voice messages to the Kofax Voice Server. In this scenario, an
additional component is required, the Voice connection.
Standalone Configuration
Owing 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 and
sent 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 Avalin
products. These applications can use Web Services to send and receive faxes via FoIP. This configuration
is recommendable for testing, troubleshooting or demonstrations, however, it is not supported for
productive use.
IP Infrastructure Requirements
This section defines the generic requirements to an VoIP/FoIP environment in order to support basic
functionality with KCS FoIP. More detailed information about compatibility, configuration and hints with
various 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 connected
via SCCP protocol to an IP environment) are not supported.
Compatibility
FoIP 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
Unicode Support
Unicode 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
Prerequisites
This section provides the prerequisites for a standard installation. The actual requirements may be
reduced (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
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restart the KCS FoIP. Note that this 9s time-out may be increased as described in the TCOSS
Configuration 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). A
V.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. Up
to 120 G.711 passthrough mode channels are supported on physical or virtual machines.
• The following CPU resources must be available.
Channel type FoIP usage [SPECInt_Rate_Base2006] per channel
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 from
a system that is similar to your hardware.
2. The maximum number of channels can be calculated with the following formula:
(CpuPerformance - ReservedForOtherProcesses)
MaxChannels=----------------------------------------------
FoipUsagePerChannel
1. The maximum number are channels are rounded under the assumption that up to 20% from total
available CPU may be consumed by other high priority processes on the same virtual/physical
machine.
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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 CPU
performance you may receive send/reception errors with TCOSS error code JO, error category
BAD_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 may
interrupt the fax call (RTP packet generation must not be deferred more than about 50ms). At least
2 cores should be reserved for optimum quality. Refer to the Environment Guide - Platform System
Manual 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.
Setup
KCS 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 uninstall
this 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 later
import 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 to
preserve its configuration, you can now import it.
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Either disable the personal firewall or click Unblock to allow the Merlin Process to access network.
After a successful installation, the following new shortcuts are added into the start menu.
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Kofax Communication Server Fax over IP Technical Manual
2. Click No to delete all files created by KCS FoIP or click Yes to keep configuration files and the fax
storage file on the disk.
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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Basic Configuration
After the initial installation, you can configure the application. If FoIP had been installed on the computer
before (and the installation directory has not been cleared or changed), the previous configuration is used
automatically.
Select the “Configure FoIP (FOIP)” shortcut from the Start menu.
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The first two tabs, General and List of Call Peers of the configuration are expanded by default. These
tabs 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.
General
In 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 and
KCS Monitor. It may be used for a better overview if multiple FoIP instances are used. It has no
functional 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 channel
configurations in TCOSS.
• Fax Storage: Use FoIP’s internal fax storage.
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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 with
registration). Select a protocol compatible with your call peer.
Note If you select an H.323 Gatekeeper (“H.323 with RAS”) you must not use
H.323 with any other active call peer.
Remote Address: The remote IP address or host name of the call peer (Gateway, Gatekeeper,
Host Registrar or CallManager) and the relevant signaling port number.
Port Note This value supports host names for H.323 only.
Authorization: The user name and password for authentication with an H.323 gatekeeper or a SIP
User ID server. With H.323, the User ID is also us55ed as endpoint name. See Endpoint
Password Name for details.
Advanced Configuration
This 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 configured
separately both for fax (T.38 and for G.711 pass through mode) and voice data. The “Diffserv/DSCP
model” 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 failed
LastError 126“ appears in the FoIP trace and the configured DSCP values are ignored. You can check
and install qWAVE with Server Manager features as shown in the following figure.
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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 or
the 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 then
expand 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 using
RTP. When using SRTP the sip message must be encrypted as well (by
using SIP via TSL/TCP) in order to protect the SRTP session keys.
MediaSecurityCryptoSuites Offers Crypto suites in outgoing SDP. All supported suites are accepted when
offered 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)
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 value
generates DMTF signals as audio tones which should work fine for testing and calls via
POTS. If the receiver expects DTMF as RFC 2833 (like KCS FoIP), then option 1 will be
required (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 are
available:
• 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 if
T.38 is supported by your environment. Modes 20 and 40 improve interoperability but
they may require better network conditions if G.711 pass through mode is used. Mode 40
is used by default because it is the most compatible mode. Mode 60 should be used only
in 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 are
available:
• 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. It
is 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 if
T.38 is supported by your environment. Modes 15, 20 and 40 improve interoperability but
they may require better network conditions if G.711 pass through mode is used. Mode 40
is used by default because it is the most compatible mode. Mode 60 should be used only
in the rare case that T.38 must be prevented for any reason. Mode 15 if fully supported
with SIP, but the fax prompt is delayed by about 5s if the fallback to G.711 happens via
H.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 encoding
which is not correctly displayed by current releases of Wireshark Network Protocol
Analyzer (tested with 1.0.7).
Note V.34 fax is not supported with G.711 pass through mode. See V.34 Fax for
support FoIP inbound/outbound T.38 modes.
RedundancyLS Redundancy for transmitted low speed T.38 packets, that is, V.21 and all non-data
events. 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.34
data. See T.38 Redundancy for details.
H.323 Signaling
This 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 required
if 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 own
IP 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 tunneling
avoids additional TCP/IP connections for H.245. It may be required for calls via NAT
router.
SIP Signaling
This 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 is
supported:
• 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 SIPS
is selected, an SSL certificate must be configured in the SSL section (as it is with
TWS, Message Connector, and others).
The difference between SIPS and TLS is that SIPS requires encryption (TLS) on all
routes 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 support
TLS. TLS provides the same level of encryption for the SIP messages between
FoIP and it call peer, but it also allows to use unsecured sip connections behind the
call peer.
FoIP up to 3.23.xx (KCS 10.0) supported TLS only without SIPS. This configuration
option is still included for unexpected compatiblitiy issues during update, but it is
depcreciated and may be removed in futures releases because it does not confirm
to RFC 5630.
SIPS is automatically enabled during update to FoIP 3.24.02 (or higher) if TLS
was enabled. This means that depriciated values are automatically converted to
compatible 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 not
specified, the gateway Host/IP is used.
LocalSipDomain SIP Domain for own number ("from" field) with outgoing calls. If this value is not
specified, 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 trusted
SIP servers to assert the identity of authenticated users.
InboundNumberSource Defines how the received inbound number is extracted from the initial SIP INVITE
message:
• 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
MulticastAddress Optional. An IPv4 multicast address. Reception via multicast address can be used
for failover and load balancing installations. See Inbound SIP Failover with Multicast
Addresses for more details.
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Value Description
MulticastPeerAddresses Blank separated list of addresses (IP[:port]) that are notified after established
multicast 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 all
RFC compliant SIP implementations.
• The SIP/SIPs URI schema includes a user information and sip-domain (user-info@sip-domain) very
similar to email addresses. The user information is usually the telephone number and the sip-domain
is 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-domain
configuration in the remote call-peer. Most implementations do the same, so that, it is not required to
configure 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 values
CalledSipDomain and/or LocalSipDomain. Note that, it is typical but not required that the same
domain is used for both values.
SSL Certificate
The 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 -----BEGIN
and -----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 be
encrypted, it will be encrypted internally.
SSL Chain Certificate Optional intermediate certificate in the certificate chain to a well-known root
certificate in PEM format (Base64 encoded, including -----BEGIN and -----END lines)
SSL Trusted CA Certificates Optional trusted CA certificates for verifying remote peer certificates. (Base64
encoded, including -----BEGIN and -----END lines). These certificates are required
only 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 the
AdvancedOpenSslConf 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 for
further details.
Local IP Address IP address of local interface used for connection to TCOSS / voice server. If empty all
local 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 then
the connection with TCOSS is only possible if at least one of the checked call-peers
is reachable. Use this feature only if an alternative FoIP Lanlink IP is configured in
TCOSS.
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 has
reached Alerting or Connected state.
Call Transfer with Hold If selected, call hold is executed prior to the call transfer. This option might be
required 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. It
is relevant for all messages posted via web-service interfaces. The ECM mode for
messages 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 detailed
trace.
Note Note: OpenH323 trace does not use the trace settings (Size of Trace
File, Number of Trace Files and Append Trace). It is always appended to
file 00\trace\OpenH323.log. It’s size is not limited and may fill up you
system 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 final
name at end of call. Names are sorted by end-time. “ActiveFax” files from a
previous abnormal termination of FoiP are deleted with the first call after FoIP
restart.
BtrTraceFiles Maximum number of binary trace files. These files are created in the FoIP trace
folder with the name “fax_####.btr” where #### is a decimal number that starts
with 0000 after FoIP startup and increments with each call. If the max. number of
trace 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 if
BtrTraceLevel is set to 18.
BtrTraceMaxFileSize Maximum size of binary trace files in kBytes. If this limit is exceeded further
binary trace output will be ignored.
Trace Location Drive and path of trace location (relative paths are based on FoIP installation
folder).
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 Configuration
The 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 is
displayed:
You can either return to the configuration screen or exit the configuration. To activate the new
configuration, use KCS Monitor to restart FOIP.
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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 FoIP
installation (such as C:\Program Files\Kofax\Fax Connector\FoIP\00) and run it at any time
later 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 the
configuration are usable only on the same Windows where the configuration has been saved.
Alternatively, a DSCP value may be set in the configuration, see VoIP/FoIP Network. Policy-based DSCP
settings have priority over DSCP settings by the application (through configuration).
Note The Windows feature, “Quality Windows Audio Video Experience (qWAVE)” must be installed (as
described 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 or
later to set the DSCP value based on the sending application executable name, source / destination IP
address, 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 data
stream.
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 specify
individual 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 executable
names 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 of
data streams.
• We strongly recommend that you configure the QoS-policy based values only in the “Computer
Configuration” 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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43
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 Windows
service.
The application provides a web portal. You can access this portal via any standard internet browser using
the following address: http://<Computer>[:Port]/
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Each line represents a single call peer. The following table provides a more detailed description of the
columns.
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 state
includes calls where:
• the remote side was busy.
• the remote side did not answer the call.
• fax transmission or reception failed for any reason.
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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). See
Supported TIFF Options for format details.
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• When a message is successfully delivered, it is moved from the Pending folder into the
Processed folder of the fax storage.
• When a message is not delivered successfully, it is moved to both Processed and Failures
folder.
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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.
The font “Microsoft Sans Serif” used in the fax header line supports characters from several different
alphabets including Latin, Latin Extended, Greek, Cyrillic, Thai, Hebrew, and Arabic (with the restriction
that Hebrew and Arabic is written left-to-right).
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Fax Storage
In the fax storage, all faxes which have been sent or received via storage are stored. Messages sent or
received via TCOSS are not saved to the storage. This storage is divided into several folders, which can
be 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 a
storage folder. Use the Refresh button to update the list.
• 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. No
processed messages are kept.
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Inbound Messages
This 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.
Click to display the content of the fax in a TIFF viewer or to save the
fax in TIFF format.
Click to display additional information about a fax message. See the
following 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 Messages
This 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 within
the 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 move the fax from the pending to the processed folder.
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Chapter 4
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).
1. Configure User Module for FoIP (UFI) channels on the TCOSS server as described in the TCOSS
Fax 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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It uses 127.0.0.1 (localhost) as gateway address and therefore does not require any FoIP environment.
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This section provides step-by-step instructions on how to install a message waiting solution with Cisco
CallManager based on KCS FoIP. See Message Wait via SIP Notify (RFC 3842) for integration with
Siemens OpenScape.
This installation assumes that the steps listed in Installation of Fax for KCS have been completed. For
details, refer to "Messages Wait Integration" of the VoIP / FoIP Environment Guide.
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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:
2. Use the mailbox access number instead of the MWION/MWOFF in the test call (Step 2) and MWION/
MWIOFF services (Step 3).
This chapter describes an installation example involving TCOSS, Voice Server and KCS FoIP v3. The
following 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 recorded
with 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, which
should 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.
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2. Set the server path and the access number. The default password for user TCVMAIL (which is
installed below) is TCVMAIL.
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5. Enter IP address and password of FoIPv3. In this example, we use 127.0.0.1 because it is running
locally.
6. After the installation, when TCOSS has been started, run C:\TCOSS\System\vmaildep.bat to
install the required services and users.
7. For a more detailed description of various setup options, refer to the Voice Server documentation.
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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 which
starts with “V”.
Example config line 254 ‘17908=V7908 ,
config line 254 ‘17908=V7908 ,
config line 255 ‘17909=V7909 ,
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 hear
a prompt saying “This is TOPCALL voice mail, please enter the number of your mailbox and press
hash”. Additionally, you should see activity in the Calls line of FoIP application in KCS Monitor:
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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.
7. Play the file again via telephone - select Tools | Use Telephone and click Play.
Your telephone should ring and the message should be played.
This section provides some hints and FoIP specific details for using USN with FoIP. It is assumed that you
are familiar with USN as described in the Voice Platform Technical Manual.
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Refer to the "Transfer the Voice Call to Fax" section in the Voice Platform Technical Manual.
If enabled, the call is transferred from Voice mode to Fax upon detection of CNG within the configured fax
detection 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 CNG
tone from received G.611 samples are not supported.
Hints
As with Line Server Model 305, all numbers with voice calls are converted as with fax calls using both the
number conversion table. Optionally, the configured PABX prefix may be used (see configuration lines 131
to 133, 2nd position of line 295).
Supplementary Services
This section describes supplementary services, such as call diversion, DTMF reception, call transfer and
more.
During call diversion for voice mail, the original called number must be delivered to the Voice Server in
the signaling protocol. When using multiple redirection, KCS FoIP may receive two or more diverting
numbers. In such a case, the configuration value Call Diversion Mode in the KCS Integration section
selects 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 SIP
The SIP protocol uses a Diversion header field. If you use a Cisco CallManager you need to enable
the delivery of this header field. Open the configuration page of your SIP trunk and enable Redirecting
Diversion Header Delivery for inbound and outbound.
Via H.323
KCS 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.
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Note PBX might send the redirecting number along with the flag national/international/subscriber
number. 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 instead
of the called party number.
This feature does not work if the redirecting number is suppressed. See Suppression of Redirecting
Number.
DTMF Reception
VoIP detects DTMF signals delivered via RTP according to RFC 2833 (RTP-NTE). This signaling can
be 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 also
detected. DTMF signaling using SIP notify is not supported.
Where <Dtmf> may contain any valid DMTF digits (0..9, #, *, A, B, C, D), P (1s delay) or Q (4s delay). You
may 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 described
in the FoIP configuration section “Fax”.
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 Transfer
Call 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 Voice
server and elects to be connected to the operator instead of leaving a voice mail to the originally called
person: behind the scenes, the KCS Voice server acts as the transferor, the caller is the transferee and
the operator is the transferred-to-party. After the call transfer is completed, the transferee (the caller) is
connected with the transferred-to-party (operator).
For information about call transfer configuration in KCS FoIP, see KCS Integration.
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 function
the 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). The
supported 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 / FoIP
Environment Guide.
Note When using call transfer with voice mail, we recommend that you enable only a single G.711
codec. Having both enabled can occasionally lead to payload mismatch. See Voice.
Configuration Hints
This chapter describes hints for some typical use cases.
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For example, when using a Cisco gateway, the calling party number can be used the select the dial-peer
which 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 user
dependent (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 channel
configuration. It is recommended to use configuration lines 276 to 279 which already contains an outgoing
number 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 the
called party number type (prefix “I”, “TI”, “TN”, “TC”, …) must be at the begin of the converted line (and not
before the called party number). An alternative configuration which is supported by FoIPv2, FoIPv3 and
LS1 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 does
not work with LS1 because it does not start with “TC”. But a conversion to TC8123%MWION;4632;4630
works.
Fax on Demand
Fax on demand is supported as described in the TCOSS System Manual for Line Server Model 305.
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Reference Configuration
This reference configuration uses KCS Tandem server, 2 FoIP instances and 2 gateways. Solid line is
used for components and connections that are used during normal operation. Dashed line is used for
components 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 during
normal 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 set
to "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 KCS
configuration. When using FoIP without fix of SPR70426, that is, 3.12.01 or older, the call-peer check
should 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 this
configuration step depends on the used gateway or IP BAPX. Refer to the VoIP / FoIP Environment
Guide for further details. An example for Cisco gateway can be found in Call-Peer Check.
The primary master is connected with FoIP #1. FoIP #1 distributes outgoing calls (using round robin) to
gateway #1 and gateway #2. Incoming calls received at any gateway are handled by FoIP #1. Both the
secondary 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 removed
fault 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 as
described in Behavior If an Incoming Call Is Rejected. The gateway will reconnect the incoming call to
FoIP #2.
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• Network connection #5 fails: All outgoing and incomings calls are done via gateway #2. Incoming
calls on gateway #1 fail because the primary IP is not reachable and FoIP #2 does not have a
connection to TCOSS. It is the responsibility of the connected telephone system that incoming calls to
gateway #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 has
a working connection to both gateways.
Failover strategies are configured via WConfig in config line 291, position 12. See Description of UFI
Specific Configuration Lines.
For each TCOSS channel, it is possible to configure which call peers are available for outgoing calls. This
is configured via WConfig in config line 291, position 9. Position 10 determines the call peers for incoming
calls. See Description of UFI Specific Configuration Lines for details.
Failover is currently not supported for fax messages sent via storage.
Call-Peer Check
The call-peer check may be activated in the KCS section of the FoIP configuration. It prevents situations
when 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 the
FoIP status overview:
The check sends a SIP OPTIONS request to all active SIP call peers that should be checked. The
connection to TCOSS is allowed if at least one checked call peer returns a SIP response value between
200 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 will
be restarted with an event log (ID 35013 - Red alert 'FoIP will be stopped because the checked call-peers
are down …)
On incoming calls (from ISDN to IP) the gateway matches the first dial peer by its destination pattern
to the incoming MSN. The dial peer 80000 will win. If the IP address is unreachable it will attempt the
next dial peer with the same destination pattern (80001) which leads to the secondary master. Please
consider 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 IOS
documentation for details).
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If multiple FoIP instances are active with the same multicast address, each one starts with the inbound
call (like a multi-ring feature). The first instance which accepts the call (by sending a 200 OK response to
the INVITE message) gets the call. This instance also informs all other instance to release it, which will
end the call with TCOSS error code “JS” and internal error number 12722 in the FoIP trace.
Configuration Steps
1. 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 not
configured 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 to
the configured multicast address. This works but causes an additional multicast packet
in the network. Do optimize the installation it is recommend to enter the Unicast IPs of
all other FoIP instances using the same multicast IP in the configuration value SIP |
MulticastPeerAddresses.
Example
FoIP 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 225.1.1.1
which 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 dialog
with 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 all
other 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.
• Lost multicast INVITE [1] or response [2]: The gateway retries the INVITE packet until it gets a
response (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 released
without further messages.
• Lost multicast peer notification [5]: FoIP #2 retries transmission for about of its response [3] until
A SIP Registrar is a server that accepts REGISTER requests and places the information it receives in
those requests into the location service for the domain it handles.
Configuration
To 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 a
registered number. See List of Call Peers.
To verify that FoIP can connect to the server, start FoIP Web Portal and click Status. You should find
similar 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 is
that fail-over is handled by the Gatekeeper in that case.
• Only one registered number is supported.
Note The Special Information Tone is a repeated series of signals with 950Hz, 1400Hz and 1800Hz sent
in that order.
If reception into storage is configured, the incoming call is always accepted. If the received fax cannot be
saved 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 (for
incoming calls) without making a real fax/voice call. If FoIP gets a SIP OPTIONS request it returns the
SIP status code shown above with the exception that the called party number is never checked. Thus
whenever FoIP is ready to handle any inbound calls it returns a success (200 OK) to an OPTIONS
request.
In this scenario, no fax server on the customer side is necessary. Instead, FoIP connects directly to a
server 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 not
blocked:
• 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 fax
client), Kofax FoIP solution has the benefit of being integrated with TCOSS (or other applications). No
additional software has to be installed on end user computers.
V.34 Fax
KCS FoIP supports the V.34 fax (ITU-T T.30 Annex F) through T.38 enabled fax relay according to ITU-T
T.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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Note V.34 fax is not used for the Fax Polling mode (used with Fax-On-Demand feature)
If you want to address many different gateways you can overwrite the IP address from the call-peer
configuration on a call-by-call base with a prefix in the destination number. This can be done using the
following number syntax:
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 received
by 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 and
operation 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 to
establish the outgoing call.
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Note The direct gateway IP address cannot be specified by the user if the default service FAX is
used, because it uses the address type fax which removes the dots from the IP address (the number
conversion for outgoing fax numbers are applied; refer to the "Fax Number Conversion" section in the
TCOSS Application Module Manual for more details). Instead, either a service with number type FREE
may be used or the gateway IP address will be added during rr99 routing.
The default behavior of NAT routers is that they support connections initiated by any host within the
private network to any host in the public network but they block connections from the public network to
any 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 simple
one-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 for
signaling and media data, where the signaling connection is used to exchange the media addresses. In
that case the public member must get the mapped public (instead of the private) address of the private
member. 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 recommended
option. 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 real
used 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 has NAT router links the internal address In network B (public), gateway
the IP address 10.0.0.10. Set the 10.0.0.10 to external address connects to FoIP using the address
parameter NatTraversalIp to its 192.1.1.10 using a static NAT rule. 192.1.1.10.
external address 192.1.1.10. Use
193.1.2.3 as Gateway IP.
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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 Name
The endpoint name is transferred as display name for inbound and outbound calls. Depending on the
used 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.323
call-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 the
same User ID for all active H.323 call-peers.
When using G.711 pass through instead of T.38 the fax modem function which converts between binary
data 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 following
cases:
• T.38 is not or insufficiently supported by the IP environment.
• Encryption of media data (using SRTP) is required.
Note Silence suppression is not supported with G.711 pass through mode.
T.38 Redundancy
T.38 redundancy is a forward error correction method that improves success rate if T.38 packets are
lost in the network. This is done by adding the content of the previous T.38 packets as redundancy. As
a consequence, the size of T.38 UDP packets (and the bandwidth) increases with redundancy. See
examples below:
Note T.38 redundancy is a trade-off between the maximum supported packet-gap and the required
bandwidth!
The generation of redundancy is configured in each T.38 endpoint. For example, if you want to handle
packet loss from your Gateway to FoIP, redundancy must be configured in the Gateway. It is not required
to use the same redundancy values on both endpoints of a T.38 session. It is also not required to enable
processing of the redundancy on the received side because the receiver automatically detects and uses
redundancy.
T.38 endpoints without support of redundancy (like KCS FoIP before KCS 9.2) silently ignore received
redundancy. This means that it is compatible with an endpoint using redundancy but it cannot recover lost
packets.
KCS FoIP supports generation and reception of T.38 redundancy information according to ITU T.38
chapter 9.1.4.1. The generation of redundancy can be configured individually for low-speed (V.21 and
non-data events) and high-speed (V.27, V.29, V.17, V.34) T.38 packets. The required network bandwidth
with 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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Note This section describes the general behavior as it would appear if you are using a connection from
FoIP to FoIP. Refer to the VoIP / FoIP Environment Guide for 3rd party specific details.
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 a
certification 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.
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KCS FoIP may optionally check if the server has a valid certificate by using the following configuration
steps:
1. Add the certificate of the remote side (or its certification authority) as SSL Trusted CA Certificates in
the 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 private
key if you reinstall FoIP on a Windows computer even if you restore the complete FoIP configuration
(SolutionConfig.xml).
Note The default certificate does not protect the SIP connection against a man-in-the-middle attack.
Refer to the TWS manual for the details on how to request a key/certificate for some well-known (and
trusted) certification authorities.
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Please enter the following 'extra' attributes to be sent with your certificate
request
A challenge password []:
An optional company name []:
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 used
as 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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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 by
installing 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 trusted
web site.
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2. Open the FoIP Web GUI and ignore the certificate warning. Click the certificate error to open the
certificate.
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3. The Certificate import wizards starts. If you want to make the trust for all users on this computer (for
example, also for services), select Local Machine as store location.
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Voice Prompts
If KCS FoIP is integrated with KCS incoming calls may be used for commands or incoming calls may
fail 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 that
matches the internal error code (as described in chapter 7.2.2) that causes call termination. If no matching
file is found the file OK.wav (for success) or Error.wav (for failure) is played. All prompt files are installed
in sub folder prompts (based on FoIP installation folder). An overview of the used KCS FoIP prompts is
shown 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 was “Bad inbound number unspecified”
detected that could not be mapped to error
reasons 42411-42415
42411.wav A fax server/routing command was entered with “Bad syntax of inbound number. For example
an incorrect syntax. invalid 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 according “The requested function is not allowed with
the security configuration in configuration line current security configuration”
237/240.
43103.wav Incoming voice call without connection to voice “Incoming call has been rejected because no
server voice 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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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 first
existing file according the list below is used:
• Config\prompts\42412.wav
• Config\prompts\Error.wav
• Prompts\Error.wav
Configuration Values
This section describes the UFI channel configuration. It also describes the tracing possibilities in FoIP and
log entry filtering.
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Note The incoming call peer is detected according to the IP address used with the
first message that indicates the incoming call. If an exact match with an IP address
configured for a call peer remote address is found, the appropriate call peer number
is used. If no call peer matches, the behavior depends on position 13 of configuration
line 291.
T.38 Traces can be activated with WConfig line editor in the UFI configuration line 242 as described
below.
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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 one
channel)
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. (State
Table 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 other
than REMOTE_BUSY or CALL-NOT-ANSWERED
04 Like 03, but include calls that fail with error category REMOTE_BUSY and
CALL-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 for
standard troubleshooting)
12 BTR trace level for outgoing calls. Like Position 11 but for outgoing fax calls.
Note The BTR trace level must be set to 256 (UFI channel specific) in the FoIP configuration to allow
activation of binary traces due to number conversion table.
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.
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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
This chapter describes the basic steps necessary to migrate from other FoIP / VoIP solutions to Kofax
Communication Server Fax over IP (FoIPv3).
The table below provides a feature compare between LS1 and FoIPv3 that excludes the following
features:
• Features supported by FoIPv3 only
• Features supported by FoIPv3 and LS1
• Features specific to the used line type (such as ISDN or VoIP protocols)
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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 these
switches 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 RFC
2833 (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 the
correct higher layer and bearer compatibility if the call is forwarded to an ISDN connection.
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This migration assumes a working H.323 fax integration with a Cisco CallManager via FoIPv3 as
described in Installation of Fax for KCS and Migration of FoIPv2 (XCAPI).
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It assumes a working H.323 fax integration with a Cisco CallManager via FoIPv3 as described in
Installation of Fax for KCS and Migration of FoIPv2 (XCAPI).
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 either
10.18.136.51 or 127.0.0.1. Enter the configured password or leave it empty if no password has been
configured.
It is not necessary to reconfigure advanced voice features, MX, LN, and GW integration. Keep the
email reader (TTS) setting unchanged.
If you are installing the same Server Package release, it is not necessary to reinstall the voicelib.
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Hints
It 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 1
from “00” to “01”.
Message waiting now uses different addressing (the order of parameters is reversed). Message wait
events / service prefixes have to be adapted (format has changed). For more information, see Step 3 –
Create Services and Events in TCfW.
Feature Comparison
The H.323 Voice feature, "G.729 audio data compression algorithm" is not supported by KCS FoIP.
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Administration
This chapter summarizes administrative information for troubleshooting Kofax FoIP and lists error
messages 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
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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
For a successfully delivered message, the Response field displays the CSI number (Called Station
Identification; for outgoing messages) or the TSI number (Transmitting Station Identification; for incoming
messages).
In case of an error, the Response field provides information about the error in the following syntax:
Error Categories
The following error categories are defined:
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Error Codes
The following table gives you an overview about the 2-digit TCOSS error codes that may be produced by
UFI. 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
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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)
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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 by
T30_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 were
received
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 were
received
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 was
expected
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 within
timeout
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 any
free 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 wrong
called 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)
50000 unspecified error in any Communication WF TNC
60000 unspecified error in any Communication WF TNC
60001 folder export failed
Connection Level
The 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 is Connection to H.323 component failed
missing
10 Local resources, such as line or socket, to initiate No connection the Gateway
a call have been allocated. A communication ISDN Establish Layer 1
media, such as physical line, with the network has
been selected
15 Any low level, such as ISDN Layer 1, connection TCP/IP Connection (or Layer 1/2 Connection) to
to a network could be established. Gateway/PBX could be established
20 Communication with network call control layer was SETUP_ACK, ALERT, CONNECT, .. message has
possible been received
25 Network connection is busy Busy from network
30 Any signal that may indicate that the remote side Busy from remote side
may be available has been received. It indicates
that the number is valid.
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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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:
This information confirms that the connection has been established and TCOSS is ready to send and
receive faxes via FoIP.
When FoIP is erroneously configured to receive faxes into local storage instead of TCOSS, the following
error is written to the log:
In this case, FoIP configuration must be changed. See the General section of Basic Configuration.
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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 are
added to the menu on the left side.
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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 correct
destination 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 Supported
TIFF Options for format details.
To send a test fax, enter the destination fax number (and possibly the other optional parameters), choose
a TIFF file that should be faxed (or leave the field blank to use a default file), and send the message
by clicking the “Send Fax” button. You might use the “Preview” button before sending to view the fax
message including the generated header line.
A message delivered in real time is not added to the fax storage. Instead, message delivery starts
immediately 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 is
displayed in the bottom frame:
T.38 Player
Note The information from this chapter and the functionality described therein is important for a short list
of 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 couple
of T.38 fax calls recorded on (for example) a customer site, it is now possible to reproduce specific fax
transmission 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.38
fax session to another fax recipient. It may contain only the transmitted (Tx) T.38 UDP packets. This
filtering 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. The
converter can filter either transmitted (Tx) or received (Rx) packets and both can be replayed using the
player.
Note If you intend to use the T.38 PCAP Converter to generate XML from the recorded traces, you can
save 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 the
Wireshark 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 destination
ports. 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 select
Displayed under Packet Range.
Initiating a Call
1. To replay the fax session, click the T.38 Player link from the menu. The following page is displayed.
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Note A full Wireshark without any extractions does not work properly. Alternatively,
you can enter a T.38 XML file.
3. Click Send Fax to start the fax transmission. A response is displayed in the lower window, which
consists of an informational table showing call ID, state and disconnect reason, and Refresh and
Disconnect buttons.
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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 Call
By clicking Disconnect at any time during an active call, the call is disconnected immediately. You can
verify this by clicking Refresh and thus updating the call information.
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If, for example, the fax recipient asked for any retransmission for any reason or even requested a different
fax 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.
PCAP files are binary files, and are complicated to read and edit manually. On the other hand, the
XML format displays the same information in a user-friendly way. The converter also allows to filter the
conversation into sent and received packets.
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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 to
relative time stamps, that is, how much time occurred since the previous
frame. (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 UDP
packets.
2. Click Start.
Test Server
1. 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 the
predefined function calls to FoIP are updated according to the selected template. The generated
URL 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 browser
window. 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 is
therefore described rather superficially.
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Chapter 8
Kofax Communication Server Fax over IP uses a web service interface for communication with fax
servers. The functions of this interface are defined in the “fax.wsdl” file. The functions are described in
detail 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 exceptions
specific 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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Specifications
Only the “bilevel” (black and white) TIFF image type is supported. Additionally, the TIFF tags listed in the
following table are restricted to the specified values:
TIFF tag Supported values
Compression (259) 1: No compression
2: Modified Huffman run length encoding
3: 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 right
margins 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-endian
and big-endian byte order of the TIFF file is supported.
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5. Confirm the configuration by clicking OK and then proceed with printing as usually. A new TIFF file
will be created.
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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
TCIMG32
The KCS image conversion library TCIMG32 is a part of TC/Link and TCfW installations. The following
TIFF outputs can be used for faxing via FoIP.
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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. The
version used in FoIP can be downloaded here: http://www.topcall.com/sources/7za442.zip.
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