User Manual

KMG One line

Table of Contents
1. About this manual 10

1. How to read this manual 10

2. Presenting the KMG line 10

2. General overview of models 11

2. KMG 200 One 11

2. Front view 11

2. Rear view 11

2. KMG 3200 One 12

2. Front view 12

2. Rear view 12

2. Features of the KMG One line 13

2. Safety features 13

2. Optional features 13

2. Telephony interfaces - KMG module 14

2. Physical connection of KMG modules 15

2. Capacity for simultaneous calls 15

3. Technical specifications 15

3. Media gateway 15

3. External telephony module 16

3. Telephony interfaces 16

4. Installing KMG 17

4. Network connection 17

4. Analog connection - FXO 20

4. Analog connection - FXS 20

4. Centronics connector 20
Table of Contents
4. RJ45 connector 21

5. Getting acquainted with the Web interface 21

5. Access computer requirements 21

5. Access computer requirements 21

5. User interface sections 22

5. Checking the KMG version 22

6. KMG operation 23

7. Configuration 23

7. Network 23

7. Configuring the network 24

7. Configuring the Firewall 25

7. Configuring intrusion detection 26

7. Managing TLS certificates 28

7. System 29

7. Configuring the operation mode 30

7. VoIP Settings 31

7. Configuring Services 32

7. Setting the date and time 32

7. Manually changing the date and time 33

7. Changing the date and time automatically using NTP/SNTP 33

7. Setting the configurations history 34

7. Telephony 34

7. Signaling 34

7. Editing the signaling profile 34

7. Copying the signaling profile 35

Table of Contents
7. Applying a signaling profile to the telephony module 35

7. Setting the properties of the E1/T1 Links 36

7. Dial Plan 37

7. SS7/SIGTRAN 37

7. SS7/SIGTRAN settings 38

7. MTP2/M2PA 38

7. Point Codes 38

7. MTP3 38

7. ISUP 39

7. Features 39

7. Integrations 39

7. Insight! 39

7. Device Management 39

7. TR-069 39

7. Other systems 40

7. Survivability 40

7. Configuration 41

7. Network 41

7. Servers 42

7. Register Authorization 43

7. Network settings 43

7. General options 44

7. Advanced options 44

7. Media profiles for Register Authorization 45

7. Servers for Register Authorization 45

Table of Contents
7. Policies for Register Authorization 46

7. All methods: User-Agent validation 46

7. REGISTER Method: pre-authorization 46

7. INVITE method: Checking the registration status 47

7. Global settings for Analytics 49

7. Routing 49

7. NAP 50

7. Creating an E1/T1 trunk 50

7. Creating an SIP trunk 51

7. Configuring GSM channels 53

7. Options 53

7. Configuring FXS channels 54

7. Registering extensions 54

7. Proxy list 55

7. Extensions 55

7. Ring pattern 56

7. Cadences 56

7. Configuring FXO channels 56

7. Configuring NAP groups - NAP GROUP 56

7. Load balancing algorithms 57

7. Channel allocation algorithm 57

7. Channel allocation: automatic mode 58

7. Channel allocation: first free channel 58

7. Round-robin (circular allocation) 58

Table of Contents
7. Sorting options 58

7. Routes 58

7. Regular expressions 60

7. Retry 60

7. Configuring call profiles 60

7. Configuring the Call Classification feature - Analytics 64

7. Configuration steps 64

7. Configuring disconnect behaviors 65

7. Configuring Analytics behaviors 65

7. Transferring audio files to the KMG 69

7. Associating the Analytics behavior to 69

a profile

7. Associating a profile to routing 70

7. Associating the classification feature 70

to the SIP trunk, E1/T1 trunk and
GSM channels

7. Setting up an operating period for routing 70

7. CDR - Call Detail Record 71

7. Customizing and activating a CDR 71

7. Sending a CDR to a RADIUS server 72

7. Sending a CDR to an FTP server 73

7. Setting up the CDR retention time in KMG 74

7. Configuring portability 74

7. Portability via Webservice 74

7. Portability via Local Database 75

7. Importing data 75
Table of Contents
7. Scripts 76

7. Adding a script 77

7. Managing SIM cards 77

7. Associating a new SIM card with a group of existing GSM 78

channels - NAP GSM

7. Adding a SIM card manually 78

7. Creating SMS routes 78

7. SMS sending test 79

7. Configuring the reception of SMS notifications 79

7. High Availability - HA 80

7. Network connection for HA 80

7. High Availability with external telephony modules 81

7. Configuring High Availability 81

7. High Availability operation 81

7. Verifying the master and the spare gateway 81

8. Monitoring 81

8. NAP 82

8. SIM cards 82

8. Devices 83

8. Status 85

8. Links 85

8. Restarting 87

8. Blocking 87

8. Channels 87

8. Monitoring GSM channels 88

Table of Contents
8. Statistical Call Chart 88

8. Device statistics 88

8. Status of the channel 88

8. Links 88

9. Diagnostics 89

9. Downloading logs 89

9. Understanding the log messages 89

9. Changing register levels and diagnostics mode 89

9. Diagnostics mode 89

9. Advanced options 90

9. Logs 90

9. Config 90

9. Licenses 90

9. OLD 90

9. CDR 90

9. Packet capture 91

9. Audit 91

10. Management 92

10. Monitoring KMG through SNMP 92

10. Gateway information 93

10. Remote Terminal - CLI 93

10. Linux Systems 93

10. Turning off the gateway 93

10. Restarting the gateway 93

Table of Contents
10. Managing system users 93

10. Changing a user password 94

10. Creating a new user account 94

10. Editing and deleting user accounts 94

10. Accessing KMG via FTP 94

10. Licenses 95

10. Provisioning 95

10. Downloading configurations 95

10. Applying a provisioning file to another KMG 95

10. Requirements 95

10. Procedure 96

10. Update 96

10. Serial port 96

10. Troubleshooting 96
 1. About this manual
This manual is directed at professionals responsible for managing media gateways of the KMG line.
This document contains the necessary information for installation, configuration and management, as well as
technical specifications of the media gateways.

1. How to read this manual

If this is your first contact with KMG, we recommend that you read this guide in its entirety. If you are
already familiar with the gateway, the main procedures are divided into topics to make it easier for you to find
answers to your problems or questions.

At the beginning of each configuration section, the route for accessing the configuration interface is
described. Depending on the version of your KMG, the access route may be different. To check which version of
KMG you are using, please refer to the "Check KMG version" section, in this manual.

Some features are only available in certain models, and this will be informed at the beginning of the
section describing the feature. This also applies to features that are made available through acquisition of additional
licenses.

2. Presenting the KMG line

KMG is a line of media gateways developed by Khomp to offer modularity and security for your
telephony systems.

Modularity because you can have all of your telephony interfaces together in a single media gateway.
This makes it possible for you to have digital E1/T1 telephony interfaces, analog FXS and FXO interfaces and
mobile 2G and 3G GSM interfaces in one media gateway.

All models in the line include the security features provided by SBC - Khomp’s Session Border Controller.
This way, you will have the features you need to increase the security of your telephony network.

In addition to the features mentioned, the media gateways of the KMG line will enable you to decrease
your costs with telephony call charges. With the creation of lower cost routes, you will be able to direct each call to
the carrier that offers the lowest call charges. In addition to lower cost routes, it is possible to create fallback routes,
balancing and routes that operate in accordance with programmed hours.

Image of KMG 3200 One model

All configuration and management tasks are performed through a Web interface that can be accessed
from any browser.
2. General overview of models
In the following you will find a general overview of the various models of the KMG line.

2. KMG 200 One

The KMG 200 is a low-density media gateway that supports up to 240 simultaneous calls with up to 2
internal E1/T1 links, as well as 8 external modules for E1, GSM, FXS, FXO or VoIP SBC technologies. All with the
possibility of up to 120 calls with transcoding (G.729/G.722 ↔ G.711). See the section "Capacity for simultaneous
calls" in this manual for more information about the capacity of the KMG 200. It can be acquired with an option of 0,
1 or 2 internal E1/T1 links.

2. Front view

Caption:
1 Power button

2 Power LED

3 Disk activity status

4 Telephony modules activity status

2. Rear view

Caption:
1 E1/T1 Link #1

2 E1/T1 Link #2

3 ETH0 to ETH4: Gigabit Ethernet ports Standard network port

4 USB 2.0 port

5 VGA port
 2. KMG 3200 One
The KMG 3200 is a high-density media gateway that can have up to 64 E1/T1 links or 2000
simultaneous calls for E1, GSM, FXS, FXO or VoIP SBC technologies. All with the possibility of up to 1000 calls with
transcoding (G.729/G.722 ↔ G.711). See the section "Capacity for simultaneous calls" in this manual for more
information about the capacity of the KMG 3200.

The telephony interfaces are made available solely through external telephony modules - the KMG Module (see the
section "Telephony interfaces" in this manual for more information).

2. Front view

Caption:
1 Power button 4 Disk activity status

2 Power LED 5 Front LCD display navigation buttons

3 Redundant power supply status 6 LCD display to show gateway information

● Off: Both power supplies are on
● Flashing: One of the power supplies is 7 Serial RS-232 port
off.
8 2.0 USB port (for use by Khomp)

2. Rear view

Caption:

Caption:

1 USB 2.0 port

2 VGA port

3 Network interfaces for management and VoIP

4 Network interfaces for connecting external telephony modules or VoIP

5 Redundant power supplies (optional)

 2. Features of the KMG One line
The following features are included in all models in the KMG line.

● HTTP/HTTPS Web interface for user management and access control

● Supports the following telephony interfaces: E1/T1, FXS, FXO, and GSM 2G and 3G
● Support to ISDN and R2 TDM signaling (R2 is available for E1 only)
● Supported codecs:
○ G.711 A-law
○ G.711 µ-law
○ G.729A
○ G.722
○ GSM
○ DVI4
● Echo canceling
● Routing based on:
○ Source and destination
○ Prefix
○ Hour
○ Allows queries to the portability database
○ Routing script
● Load balancing and routing
● Customizable CDR and RADIUS support
● SNMP support
● Jitter Buffer to automatically correct network delays in a dynamic manner
● NAT traversal (external IP, STUN)
● Provisioning (settings export and import)
● TR-069 Support
● SIP-I Support
● DTMF: In band, Out band – RTP (RFC 2833) or Out band – SIP Info
● History and restoration of changes to settings via web
● Remote terminal with advanced CLI (Command Line Interface)

2. Safety features
● Network topology hiding
● Detection of malicious RTP packets
● Access Control List - ACL
● Protection against malformed packets
● Call blocking by destination and source
● Protection against DoS/DDoS attacks
● SIP TLS and SRTP protocols (SDES and DTLS)

2. Optional features

The following features are available through acquisition of additional licenses (not included in the standard KMG)

● Survivability - SAS
● High Availability - HA¹
● Analytics - Call Classification Feature
● Register Authorization
● ISUP (SS7), SIGTRAN Support

¹Resource available in all models, however there is no HA support for the internal model of the equipment, if one is present
 2. Telephony interfaces - KMG module
Telephony interfaces can be made available through the KMG Module - an external telephony module or
one that is integrated in the media gateway (KMG 200 and KMG 400 ).

There are different models of KMG Modules that offer E1/T1, FXS, FXO and GSM interfaces, providing a
scalable media gateway with optimized capability for expansion.

Images of KMG Module

The modules are connected to the KMG server through an Ethernet interface. They were designed to be
installed in a rack, with each module occupying 1U and a half 19-inch rack.

Example of installation of a KMG 3200 with 8 KMG Modules

Back view of KMG Modules

KMG E1 Module KMG GSM Module

KMG FXO Module KMG FXS Module

KMG Modular Module - with telephony interfaces

The KMG module is a commercial variation of the EBS module. It is exactly the same
Note hardware, differing only in the way features are licensed.
 2. Physical connection of KMG modules
The modules must be connected to the interfaces that will be used. Additional information can be found
in the section "7.1.1 - Network configurations". A physical connection may be made as shown in the image below:

The KMG module is connected to the configured network interface, and if the number of external
modules is greater than the number of ports available in the server, the module’s second network interface can be
connected to the first network interface of the second module, making it possible to use the maximum capacity of
the equipment with a chain connection of up to one level.
If an HA solution is being used, the second telephony module can be connected to the spare KMG
server, establishing a complete high availability connection in the system, involving all of the external telephony
modules.

2. Capacity for simultaneous calls

The number of KMG simultaneous calls is the sum of all telephony channels (E1/T1, FXS, FXO, and
GSM) and VoIP sessions. If the equipment is configured in transcoding mode, this number is reduced by half when
using codecs g.729 and G.722 due to the processing that is performed.
KMG 200 - Up to 240 simultaneous TDM or VoIP (SBC) calls. This number is reduced by half when transcoding,
resulting in 120 simultaneous calls using any signaling.
KMG 3200 - Up to 2000 simultaneous TDM or VoIP (SBC) calls. This number is reduced by half when transcoding,
resulting in 1000 simultaneous calls using any signaling.

3. Technical specifications
The following lists the technical specifications for the media gateway and the telephony modules.

3. Media gateway
Here are the technical specifications for the media gateway.

Item KMG 200 One KMG 3200 One

Network 5 x RJ45 10/100/1000 Mbps 13 x RJ45 10/100/1000 Mbps

16.9" x 1.7" x 7.3" (430 mm x 44 17.2" x 1.75" x 14.9" (437.8 mm x

Dimensions (W x H x L)
mm x 185 mm) 44.45 mm x 380 mm)

Approximate weight (without

7.05 lb (3.2 Kg) 16.5 lb (7.5 Kg)
packaging)

Operating temperature 32–122 ºF (0–50 ºC)

Operating humidity 10–90% non-condensing

Power supply - Full range 100–240 VAC, 50/60 Hz

Maximum power consumption 150 W 120 W

3. External telephony module

Here are the technical specifications for the KMG Modules.

Item KMG E1 Module KMG GSM KMG FXO KMG FXS Modular KMG
Module Module Module Module

Network 2x RJ45 10/100 Mpbs

Dimensions (W x H x L)
8,68" x 1.75" x 11.02" (220.5 mm x 44.5 mm x 280 mm)

5,73 lb to 4.4 lb to
5,73 lb to 6,17 lb
Approximate weight (without 8,81 lb 4.85 lb (2 5.9 lb (2.7
(2.6 Kg to 2.8 Kg) 5.7 lb (2.6 Kg)
packaging) (2.6 Kg to Kg to 2.2 Kg)
4 Kg) Kg

Operating temperature 32–122 ºF (0–50 ºC)

Operating humidity 10–90% non-condensing

Power supply - Full range

100–240 VAC, 50/60 Hz

Maximum power consumption

150 W 150 W 120 W 150 W 120 W

3. Telephony interfaces
E1/T1

Item Description

Connector available Coaxial BNC or RJ45

● BNC Coaxial: 70 Ohms

Electrical resistance
● RJ45: 120 Ohms

Signaling ISDN and R2 (R2 for E1 only)

Use of ISDN PRI (Primary Multiplex)

GSM
Item Description

Size of SIM card supported mini SIM (2FF)

● 2G quad-band: 850/900/1800/1900 MHz

Band supported ● 3G Penta-band: 850/900/1700/1900/2100 MHz with fallback to
2G quad-band
FXO

Item Description

Minimum ring sensor 13.5 Vrms@ 13–68 Hz

Off-hook sensor 5–20 V

FXS
Item Description

Ring voltage 50–70 Vpp/25 Hz

Line type Balanced

Loop resistance (maximum) 1300 Ohms, including telephone set

Loop current 20 mA

Extension power supply 27 VDC

● Up to 3.1 miles (5 km) using

100 Ohms/Km wire (AWG24 copper
conductor)
Supported distance:¹
● Up to 2.48 miles (4 km) using 125
Ohms/Km wire (AWG26 copper
conductor)

¹ Considering a telephone set with a 300-Ohm-loop circuit.

4. Installing KMG
KMG was designed to be installed in 19-inch racks. The media gateway (server), occupies 1U. The
external telephony modules - KMG Module, occupy an additional 1U and a half rack. If you acquire only one KMG
Module, a tab is included for installation in a rack.

Installation, both of the media gateway and the KMG Module, should be done in environments with a
temperature of between 0 to 50º Celsius and humidity between 0 to 90% without condensation.

4. Network connection
The media gateways have a switch with a dedicated network controller. The number of external network
ports of this switch varies depending on the model. The following table lists the description of the gateways and
their number of network ports.

Number of ports Model

KMG 200 One

5
KMG 400 One

8 KMG 1600 One

13 KMG 3200 One

KMG with 5 network ports - KMG 200/400

Rear view of KMG 200 One with two E1 Links

The network interfaces of the gateways that have 5 network ports are called eth0, eth1, eth2, eth3, and
eth4. These five interfaces have the same MAC address, as they are connected to the same port on the network
controller. Therefore, it is necessary to connect each KMG port to a physically separate network. For each virtual
network interface, a VLAN can be configured to allow propagation to the local network.

In the following image, KMG is connected to 5 physically separate networks.

Network internal flow in the 5-network port KMG

Connect the KMG ports to different physical networks, since the MAC address is shared between the
network ports, as shown in the figure below.

Example of correct installation Example of incorrect installation

KMG with 8 network ports - KMG 1600

The KMG 1600, despite having 8 external network ports, has the same specifications and connections
as the KMG 400.

KMG with 13 network ports - KMG 3200

Rear view KMG 3200 One

The network interfaces of the KMG with13 ports are arranged in the following manner: The eth0, eth1,
eth2, eth3 and eth4 ports can be used exclusively for the network and have the same MAC address, because they
are connected to the same port as the network controller. Therefore, ports eth0 to eth4 must be connected to
physically separate networks.

The last 8 network ports, from P0 through P7, are specified for connection to external telephony
modules - KMG Module, but they may also be used for connection to other VoIP networks, provided that they are all
different. These interfaces have their own distinct MAC address different than that of ports eth0 through eth4.

In the image below, the 13-port KMG is connected to 2 physically separate networks.

Network internal flow in the 13-network port KMG

Example of correct installation Example of incorrect installation

 4. Analog connection - FXO
The analog FXO interface is made available through an RJ11 connector. The following table gives the
cabling schematic for connection to the FXO interface.

RJ11 Pin Function

1 Not connected

2 Not connected

3 Ring

4 Type

5 Not connected

6 Not connected

4. Analog connection - FXS

The FXS interface can be made available in two ways: Centronics Connection.and RJ45. The KMG FXS
module uses a Centronics connection. In the KMG 400 and the KMG Modular module with FXS, this interface is
made available with RJ45 connectors.

4. Centronics connector
With this connection, the exchanges are connected through a 50-pin Centronics connector. The
connector must be assembled according to the following specifications in order for the module to operate correctly:

Arrangement of channels in the Centronics

connector

Pin Signal Pin Signal Chan Pin Signal Pin Signal Chan Pin Signal Pin Signal Chan
nel nel nel

1 TIP 1 26 RING 1 1 1 TIP 1 26 RING 1 1 1 TIP 1 26 RING 1 1

2 TIP 2 27 RING 2 2 2 TIP 2 27 RING 2 2 2 TIP 2 27 RING 2 2

3 TIP 3 28 RING 3 3 3 TIP 3 28 RING 3 3 3 TIP 3 28 RING 3 3

4 TIP 4 29 RING 4 4 4 TIP 4 29 RING 4 4 4 TIP 4 29 RING 4 4

5 TIP 5 30 RING 5 5 5 TIP 5 30 RING 5 5 5 TIP 5 30 RING 5 5

6 TIP 6 31 RING 6 6 6 TIP 6 31 RING 6 6 6 TIP 6 31 RING 6 6

7 TIP 7 32 RING 7 7 7 TIP 7 32 RING 7 7 7 TIP 7 32 RING 7 7

8 TIP 8 33 RING 8 8 8 TIP 8 33 RING 8 8 8 TIP 8 33 RING 8 8

 4. RJ45 connector
The FXS interfaces of KMG platform are provided in 8-channel modules, divided into two RJ45
connectors. It is possible to use an J45 ↔ RJ11 conversion box, or to connect directly to the analog devices using
the following scheme.

RJ45 Standard color Connecto Connector

Pin TIA568A r 0–3 4–7

1 Green-white RING 1 RING 5

2 Green TIP 1 TIP 5

3 Orange-white RING 2 RING 6

4 Blue TIP 3 TIP 7

5 Blue-white RING 3 RING 7

6 Orange TIP 2 TIP 6

7 Brown-white RING 4 RING 8

Female
8 Brown TIP 4 TIP 8 Connector

5. Getting acquainted with the Web

interface
This section provides a description of the KMG Web interface. The Web interface allows for
"Configuration", "Monitoring", "Diagnostics" and "Management" of the products in the KMG line.

5. Access computer requirements

The client computer used for accessing the KMG Web interface must meet the following requirements:

● Network connection with KMG.

● One of the following browsers:
○ Google Chrome.
○ Mozilla Firefox.
○ Internet Explorer.
● Recommended screen resolution: 1920 x 1080 pixels.

5. Accessing the Web interface

The following procedure describes how to access the KMG Web interface.

1. Open a standard Web browser.

2. Enter the IP address attributed to the eth0 of the KMG. If the IP address is in the factory configuration,
enter the IP address described in the following table, according to the model of your KMG.
 KMG Model Description Subnet Mask

KMG 200 One 10.10.10.10

KMG 400 One 10.10.10.20

255.0.0.0
KMG 3200 One 10.10.10.30

KMG 1600 One 10.10.10.40

3. Enter your user name and password. If this is your first access, use the user name and password
described as follows.

User Password

admin khomp

During the first access, we recommend you change the admin user password as a security
Note measure.

5. User interface sections

The Web interface is divided into four sections:

● Configuration: Contains the options for network configuration, creation of SIP trunks, E1 trunks,
configuration of analog interfaces, GSM, codecs and other gateway configurations.
● Monitoring: Provides options for monitoring telephony interfaces and VoIP channels.
● Diagnostics: Features options to display KMG main operational messages, and enables users to
download and view Logs.
● Management: Features options to manage the gateway, such as updating the equipment version,
changing the Web interface access password, and provisioning.

Some features and options may only be available in certain versions of the equipment. This manual lists
the versions where different features are available.

The Web interface offers help for almost all fields indicated by a "?" symbol, which can help with device
configuration.

5. Checking the KMG version

The version of the KMG is displayed in the management menu under the gateway information button. A
variety of information is provided in this window including the version of the package that is installed in the KMG.
 6 KMG operation
The KMG can perform control and routing of calls coming from one network, forwarding them to
another network, according to programmed rules.

To configure routing in the gateway, you need to know the concepts of "NAP", "Routes" and how they
relate.

"NAP" (Network Access Point) is an entry and/or exit point for calls. It represents one or more E1/T1,
GSM, FXS, and FXO channels, a SIP host/user, or a SIP Trunk, thus defining groups of channels that may be used by
one or more routes as a source (incoming calls) or a destination (outgoing calls).

"Routes" are associations between source and destination NAPs. i.e., to create a route, the user needs to
select the source NAP, which is the entry point for calls in the Gateway, and the destination NAP, which are the
outgoing calls.

Routing can be based on the source number, destination number and the result of the portability query.
You can also restrict routes by schedules, set up alternative routes that will be used as "fallback" and "retry", as well
as create advanced call routing scripts using the LUA language.

When there is an incoming call, all routes that have the NAP in which the call was received as the
Source NAP will be evaluated in the priority order set in the configuration. The first route that satisfies all filters will
be used to allocate the outgoing call using the configured Destination NAP.

The outgoing call progress is monitored and may result in the "retry" behavior if it disconnects
prematurely (before the call is answered), or in the "time-out" behavior if completion occurs and the cause for
disconnecting is mapped to the incoming call profile as "retry". Fallback is a "retry"-like feature, available only for SIP
destination NAPs with multiple configured proxy servers. This feature allows KMG to change the "proxy" of the
outgoing call, if it does not respond in a timely manner, without triggering the "retry" feature.

If a "retry" occurs, route evaluations will be resumed from the next priority set for the valid Source NAP
routes.

In order to make product configuration and management easier, the Web interface was developed,
gathering all settings and features, which were distributed into 4 major categories: "Configuration", "Monitoring",
"Diagnosis", and "Administration".

To create a basic route simply complete the following steps:

● Step 1: Create a NAP that will act as the Source NAP and another that will act as the Destination NAP at:

"Interface Web" → "Configuration" → "Routing" → "NAPs"

● Step 2: Configure a Route from one NAP to another at:

"Interface Web" → "Configuration" → "Routing" → "Routes"

Details of the KMG operation and a description of the fields in each menu are provided throughout this
manual.

7 Configuration

7. Network
The menu Network contains all of the configurations pertaining to the network configurations of the
equipment. Interfaces, Firewall and TLS certificates. It is not necessary to reboot the equipment to apply these
configurations.
 7. Configuring the network
The menu Network contains all of the configurations pertaining to the network configurations of the
equipment. Interfaces, Firewall and TLS certificates. It is not necessary to reboot the equipment to apply these
configurations.

"Configuration" → "Network" → "Interfaces"

This menu displays options for configuring the KMG network interfaces. In this Web page you can
configure the network interfaces for connection of the KMG to a VoIP network, define the network interface to be
used with an external telephony module - KMG Module, as well as perform configuration of DNS and static routes.

The "Network" panel displays a table for configuration of each network interface. The following table
describes how to use each option.

The network interface options will be displayed according to the defined operation mode. Other options will be
displayed only if the settings mode is "Static".

With the edit button you can perform configurations related to the particular network interface. With the
add button you can add a VLAN interface.

See the following table for a description of each interface parameter:

Field Description

List of available network interfaces in the KMG. The number of interfaces

Interface
displayed varies in accordance with the KMG model.

Selects the operation mode for the interface:

● Disabled: Disables this interface.
● Network: Mode of operation as network interface.
Operation mode ● EBS Module: The network interface will be used for
connecting an external telephony module - KMG Module.
Display of this option varies according to the network
interface and KMG model.

Network

Field Description

Type IPv4 or IPv6 protocol that will be used in this network interface.

Select the configuration method for the network:

● Static: The IP address, subnet mask and network gateway are
assigned manually.
Configuration method
● Dynamic (DHCP): In this interface, the network definitions will
be attributed by a DHCP server that is available on the
network. DHCP over IPv6 is not supported.

IP address IP address assigned to the network interface.

Subnet Mask Subnet mask assigned to the network interface.

Definition of the network gateway. If there is another network interface

Gateway with another network gateway defined, you must specify the default
network gateway by selecting the checkbox next to it.
 It is possible to create VLANs configuring the required ID for each network interface. When add is
clicked, the following options will be displayed:

Field Description

ID VLAN ID to be used by this network interface.

Configure DNS
Configure the search domain and the addresses of the primary, secondary, and tertiary DNS servers that
will be used, according to the provided order of preference, for network name resolution.

You can define up to 3 DNS servers.

Configure static routes

You can configure static routing through KMG by providing the IP address, subnet mask, gateway, and
network interface. The static routes created have priority over other existing routes.

To enter another route, click on Add route, just below the table of static routes.

7. Configuring the Firewall

The KMG access control settings and the network services that will be able to communicate with the
gateway are defined.

The following protocols will be available, regardless of the configurations applied in KMG firewall:

● UDP for SIP, RTP, and SNMP protocols.

● TCP for FTP 1, HTTP, and SSH 1 protocols.
● ICMP. Only to respond to echo.

Access through SSH and FTP protocols has active protection against brute force attacks. Both accept a
maximum of 3 consecutive unsuccessful connection attempts before entering into a holding mode for 120
seconds. It is necessary to wait for 120 seconds before new access attempts may be accepted, and this time
interval begins to recount any time a new attempt is made before the 120-second wait time is completed.
Configuration of the firewall is performed in the following order:

1. Define a list of Authorized Addresses. When an IP address gets authorized, the other settings do not get
applied, even if they have been configured.
2. Block the addresses that are not on the list of Authorized Addresses.

You can also:

● Enable/Disable network services for each interface available in KMG.

● Configuring intrusion detection

If the "Authorized Address" field is set to "ANY", the options mentioned earlier will not be effective even if
they have been configured.

Allowed addresses
IP addresses that are allowed access to the KMG are specified. If no value is entered, it is assumed that
any IP address can access the KMG. However, access is restricted to the Web interface and SIP registration only.
This setting is used in conjunction with the Blocked Addresses setting. Click on the "Add" button and apply one of
the following options:

● ANY: Makes it explicit that any IP address can access KMG. Whether it is through the Web interface,
FTP, SSH and SNMP.
● IP address: Only the specified IP addresses can access KMG. For example: 192.168.0.5.
● Network prefix: Only IP addresses belonging to the specified range can access KMG. For example:
192.168.109.0/24.

We recommend that you only allow the IP addresses that will have access to KMG. To do this,
Note be sure to check the "Block all addresses that are not in the Authorized Addresses list" option.

Blocked addresses
When enabled, this option blocks all IP addresses that are not in the "Authorized Addresses" list.

When this option is enabled, make sure you have at least one IP address authorized to
Attention access KMG

Network Interface-Enabled Services

Here you can determine the network services that will be available for each network interface. As a
reminder, the configuration is cumulative, i.e., the list of allowed and blocked addresses takes precedence over this
configuration.

7. Configuring intrusion detection

KMG has the ability to temporarily block communication with an IP address from a device that
attempted access and was unsuccessful. Blocking is performed in accordance with the security rules defined in this
configuration section.

To configure intrusion detection using the KMG Web interface:

1. To enable this feature, click on + Intrusion Detection, as shown in the following figure:

2. Then, simply enable the type of detection desired. There are four types of detection:

● SSH access by brute force: Blocking due to an excess of attempts with SSH access failure.
● WEB Interface access by brute force: Blocking due to an excess of attempts with failure in accessing
the Web interface.
● SIP registration by brute force: Blocking due to an excess of attempts with registration failure. Available
only with an active license for Register Authorization.
● SIP flood: Protection against SIP flood. Available only with an active license for Register Authorization.

3. When you enable a type of detection, the following fields are displayed. Configure them in accordance
with your security policies:

● Maximum number of attempts: The maximum number of times that a device can attempt to access
KMG. If this limit is reached, the device IP address will be blocked for the amount of time set in the
"Blocking Time" field.
● Time interval: The interval of time, in seconds, that a device can attempt to access KMG. For example: If
there are 3 unsuccessful access attempts in 60 seconds, the IP address will be blocked.
● Block time: Time, in seconds, during which a device that exceeds the previous limits will not have
access to KMG. At the end of the blocking period, the device may try to access the KMG again.

4. Click on the "Save" button. Finally, click on the Apply button for the changes to take effect.
 7. Managing TLS certificates
"Configuration" → "Network" → "TLS Certificates"

You can register private key, certificate, and trust store sets to be used by the device's VoIP interfaces. A
private key can be generated automatically by the device along with a self-signed certificate using the "Create
self-signed certificate" button, or existing files can be sent to a new configuration using the "Add existing certificate"
button and then using the edit button of the setting you wish to change.

Only Survivability and Register Authorization modules allow configuration by interface; the
Note routing module uses the first configuration available on all interfaces.

Creating a Self-Signed Certificate

Fill out the form that will be displayed. When submitting the form, wait for the Web interface to confirm
the creation of the certificate and open the editing form.

The self-signed certificate creation form contains:

Field Description Releas

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Name TLS configuration name

Size of the private key (in bits) that will be generated using random values.
Size (bits) The larger the key, the safer it is, however, it will require more processing in
order to be used

Enables encryption of the private key with the desired algorithm; requires
Encryption algorithm
"Encryption Password"

The password that will be used for encrypting and decrypting the private
Encryption password key must be provided only if an "Encryption algorithm" has been selected

It must contain the same password entered in the "Encryption Password"

Confirm encryption
field, it should only be filled in if an "Encryption Algorithm" was selected
password

Country (C) Country informed in field C of the certificate

4.0.2

State or province (ST) State or province informed in field ST of the certificate

City or location (L) City or location informed in field L of the certificate

Organization or company Name of the organization or company informed in field O of the certificate
(O)

Organizational unit or The name of the organizational unit or department to be informed in the
department (OU) field OU of the certificate

Common name (CN) Name informed in the field CN of the certificate

E-Mail Address E-mail address of the contact person or the person responsible for the
(emailAddress) device informed in the field emailAddress of the certificate

Expiration time (days) Certificate expiration time provided in days

Add or Edit Existing Certificate
When requesting to add an existing certificate, the system will generate and save the configuration with
predefined values, and then open the editing form for changing and sending the files that you wish to associate with
this configuration.

The editing form contains:

Field Description Releas

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Name TLS configuration name

When configured, it is the password used to decrypt the private key and
will display a link to change the password. If it is not already configured, it
will display the field to add a new password.
Encryption password
ATTENTION: The private key is never changed using this field. If
you wish to decrypt and re-encrypt the key with a new password, you need
to download the key, make the changes, and upload the new key with the
new password.

If enabled, the default system trust store must be loaded and used in
conjunction with the "trust store" provided by this configuration. The
Enable Default Trust
default trust store contains the widely public certificates of certifying
Store
authorities and is provided as part of the KMG update file. It cannot be
changed.

This option depends on whether the Trust Store and/or the Default Trust 4.0.2
Store are enabled and will require verification that the certificate provided
Enable Certificate by the remote site during the TLS handshake was issued by any of the
Verification known certifying authorities. If a Trust Store is not configured and the
Default Trust Store is disabled, this setting is ignored and any certificate
offered by the remote site will be accepted.

When there is no associated private key, this field will give the option of
sending a private key. When there is a private key, it will show the condition
Private key
of use of the key and its size (if the condition is OK) and buttons to
download or remove it.

When there is no associated certificate, this field will give the option of
Certificate sending a certificate. When there is a certificate, it will show the conditions
of use of the certificate and buttons to download or remove it.

When there is no associated trust store, this field will give the option of
Trust Store sending a trust store. When there is a trust store, it will show the
conditions of use of the trust store and buttons to download or remove it.

7. System
This menu groups all of the system configurations that require rebooting of the equipment for validation.
 7. Configuring the operation mode

The mode of operation defines the operating mode and consequently the set of codecs supported by
the KMG. Configuration of the mode of operation will depend on the licenses applied in the KMG, the maximum
number of simultaneous calls supported by the KMG, and the codecs used by the trunks connected to the gateway.

To configure the operation mode:

1. Access the menu "Configuration" → "System" → "Mode of operation".

2. Configure support for codecs by selection one of the options below:
● Bridge: The KMG doesn’t apply any processing to the calls that travel through the gateway, thus allowing
any audio and video codec to be routed.
● G711: The KMG accepts calls from the following codecs: G.711 A-law, G.711 µ-law, GSM, and DVI4.
● Transcode: The KMG accepts transcoding of calls from the following codecs: G.729A and G.722.

3. It is possible to select the number of TDM channels to be used in the panel "Between TDM and VoIP", or
select the option for exclusive use in SBC mode, as explained below:
● SBC ONLY: The KMG does not accept calls between TDM and VoIP; only SBC mode is enabled.
● E1-G711: The KMG accepts calls between TDM and VoIP. There are different E1 levels registered for
each type of equipment.
4. The panel "Between VoIP and VoIP" displays the number of calls remaining according to the options that
were previously selected giving the maximum number of VoIP calls available and in accordance with the
support provided for the codecs configured.
5. Click on the "Save" button.
6. Restart the gateway for those settings to take effect. A message will be displayed with a button to
restart the gateway, or you can restart it later (see the section "Turn off and restart the KMG").

The fields "Maximum number of TDM channels" and "Maximum number of SBC calls" display
Note the capacity for simultaneous calls supported by the KMG according to the mode of operation
defined.
 7. VoIP Settings

The mode of operation defines the operating mode and consequently the set of codecs supported by
the KMG. Configuration of the mode of operation will depend on the licenses applied in the KMG, the maximum
number of simultaneous calls supported by the KMG, and the codecs used by the trunks connected to the gateway.

"Configuration" → "System" → "VoIP"

In this tab you can change the VoIP settings for the KMG. The configurations applied here are global,
applicable in all sections and SIP trunks.

It is possible to configure the ports used for each interface separately.

The following table shows the description of KMG VoIP configuration fields:

Field Description Releas

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Maximum value of the RTP port that should be used to transmit the audio
Highest RTP port
of the SIP connections to each network interface.
-
Minimum value of the RTP port that should be used to transmit the audio
Lowest RTP port
of the SIP connections to each network interface.

SIP Port used for the UDP protocol. Any server that needs to connect to
UDP Port
KMG should use this port for SIP protocol traffic to each network interface.

SIP Port used for the TCP protocol. Any server that needs to connect to
TCP Port 4.1.1
KMG should use this port for SIP protocol traffic to each network interface.

SIP Port used for the TLS protocol. Any server that needs to connect to
TLS port
KMG should use this port for SIP protocol traffic to each network interface.

Some characters have a special meaning on the A and B numbers (such

Enable automatic
as "#", for example). And need to be coded as special entities. However,
replacement of special
some devices do not accept this coding. On most cases the default
characters
configuration is the best choice.

In some configurations, the desired call source information is not in SIP

Ignore the "Contact" field
"Contact" field, but in the "To" field, instead. By selecting this configuration,
of the headers, using the
the call source data will be read from the "To" field instead of from the
"To" field instead
"Contact" field.
-

If enabled, the Header Require:100rel will be sent on messages 101-199

Use PRACK (RFC 3262)
requesting the Provisional ACK in response.

Enable RTCP If enabled, uses RTCP to control the RTP packets.

Payload type for the

Determines what type of DTMF payload will be sent on the SDP.
telephone-event
 Field Description Releas
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URI User identification Adds the selected identification to the URI.

Send the URI "transport"

Inserts the transport parameter, with the type of protocol used in the URI.
parameter

Audio packet size Determines the size of the audio packet used in milliseconds.
-
ToS for SIP packets
Signaling packet marking for ToS.
(DSCP)

ToS for media packets

Media packet marking for ToS.
(DSCP)

TLS Encryption Selects the TLS certificate option to be used by the system.
In the "Timers" section, you will find the specific SIP timing settings.

Field Description

Maximum time, in milliseconds, for retransmission of SIP packets. The

Timer H
default time is 32,000 milliseconds.

Maximum time, in milliseconds, for dialog establishment confirmation

Waits for the ACK after
(ACK) (200 OK response for an INVITE). The default time is 32,000
2xx
milliseconds.

Whenever there are changes to the VoIP configuration, the KMG must be restarted after
Note clicking on the "Apply" button.

7. Configuring Services

"Configuration" → "System" → "Services"

The services available on the KMG are configured. Currently it is possible to configure the HTTP port
and enable HTTPS service defining the port and certificate to be used.

7. Setting the date and time

The date and time in KMG can be configured manually or retrieved by an SNTP/NTP server.

We strongly recommend that the date and time of the KMG be correctly set, as the logs, the
Attention call records on CDR files, and the time-restricted routes are affected by this setting.
 7. Manually changing the date and time

To configure the date and time manually through the Web interface of the KMG:

Go to the "Configuration" → "System" → "Date and Time" menu

1. In the "Date" and "Time" fields, enter the date and time according to the location where KMG is installed
(e.g.: 01/05/2018).

an image will go here

1. In the "Time zone" field, select the time zone according to the location where KMG is installed.
2. Click on the Save button. If no other configuration is performed, click on the Apply button.
3. Restart the gateway for those settings to take effect. A message will be displayed with a button to
restart the gateway, or you can restart it later (see the section Turn off and restart the KMG).

7. Changing the date and time automatically using

NTP/SNTP

To configure the date and time manually through the Web interface of the KMG:

Go to the "Configuration" → "System" → "Date and Time" menu

1. Enable the NTP/SNTP option.

2. In the "Maximum wait time" field, set the time (in seconds) for attempting to establish communication
between KMG and the NTP/SNTP server.
3. In the "Servers" field, configure the IP address or FQDN of the NTP/SNTP server. To add another server,
click on the "Add server" option.
4. Click on the "Save" button. If no other configuration is performed, click on the "Apply" button.
5. If the date and time are changed, restart the gateway for these settings to take effect. A message will be
displayed with a button to restart the gateway, or you can restart it later (see the section "Turn off and
restart the KMG").

● It may be necessary to configure the DNS in KMG so the automatic configuration

of date and time will work properly.
Note
● Some browser versions have inconsistencies when manually updating the
system’s date and time.
 7. Setting the configurations history
"Configuration" → "System" → "Configuration history"

This menu contains the entire history of changes to configurations of the equipment. It makes it
possible to see the user and access IP, and the time of the change. It also allows for verification of what changes
were made and for undoing the changes, if necessary.

When restoring the configuration to a determined point, all of the configurations will be
returned to their prior state at the point selected. All changes made after that point will be
Attention undone.

7. Telephony
This menu displays all of the configurations related to telephony.

The E1/T1 interface must be connected to the KMG for this menu to be displayed.
Note

7. Signaling
"Configuration" → "Telephony" → "Signalization"

It is possible to change the signaling profile of the equipment that is applied to the E1/T1 links and GSM,
FXO and FXS telephony channels. All of the default configurations refer to the standard protocols. You can edit or
copy a profile to change its properties accordingly.

The profiles are applied in all telephony links or channels of the telephony module - KMG Module. It is
not possible to apply a signaling profile in only some of the channels in the same module.

7. Editing the signaling profile

Change these configurations only when necessary and if you are knowledgeable about each
Attention type of signalization.
 1. In the profile that is to be changed, click on the icon
2. Change the necessary properties.
3. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Copying the signaling profile

If you have two telephony modules that require different profiles, you can copy a profile and change its
properties. This way you create a new profile to apply in different modules.

1. In the profile that will have its properties copied, click on the icon
2. In the field "Name" define a different description from that of the original profile. Change the necessary
properties.
3. Click on the "Save" button.
4. Consult the section "Applying a signaling profile to the telephony module" and then apply the profile
created in a telephony module.

7. Applying a signaling profile to the telephony module

After creating a new profile by using a copy of an existing profile, it is necessary to apply this profile to
the telephony module. If you only edited an existing profile, this step will not be necessary, because it will be loaded
after clicking on the "Apply" button.

1. Access the "Configuration" → "Device" menu.

2. An image of a KMG Module with its properties will be displayed. Click on the module that contains the
telephony interface that will have its profile changed. The internal modules available in the KMG 200 and
the KMG 400 will also be displayed with an image of the KMG Module

3. In the column "Profile", select the profile that you created. Only profiles compatible with the technology
of the telephony interface will be displayed. For example: If the interface is an E1/T1, only signaling
profiles that are compatible with this interface will be displayed, such as R2 and ISDN, among others.

4. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.
 7. Setting the properties of the E1/T1 Links
"Configuration" → "Telephony" → "E1/T1 Links"

It is possible to change the signaling profile of the equipment that is applied to the E1/T1 links and GSM,
FXO and FXS telephony channels. All of the default configurations refer to the standard protocols. You can edit or
copy a profile to change its properties accordingly.
The profiles are applied in all telephony links or channels of the telephony module - KMG Module. It is
not possible to apply a signaling profile in only some of the channels in the same module.

The following table contains the description of the fields available in this Interface:

Field Description

Serial Enter the serial number of the E1 module.

Description of the module to aid in identification. For example: The name of the telephony
Name carrier that the link is connected to. This name is displayed during the creation of the E1
trunk (menu Configuration → Routing → NAPs).

Signaling Select the signaling of the link.

Determine whether the telephony interface should receive or generate the reference clock
in the link. There are two options:
Synchronism (clock) ● Receive: Used when the KMG is connected to a fixed telephone carrier
(PSTN).
● Generate: Used when the KMG is connected to a digital PBX.

Minimum number of digits for the destination number to be requested from the remote
Incoming digits
exchange.

If enabled, KMG will use the CRC resource. The device connected to KMG must also have
CRC
CRC support.

The configurations in the table seen previously are the only definitions that are usually changed.
Because they depend on the type of connection, PBX digital or PSTN.

For more link configurations, click on the Advanced Configurations button. More columns in the table
will be displayed. These configurations should be changed only if there is a need or if requested by your fixed
telephony operator.

The following table shows a description of the available fields:

Field Description

Operation mode Select whether the link will operate as E1 or T1.

 7. Dial Plan
"Configuration" → "Telephony" → "Dial plan"

The Dial Plan determines an expected sequence of dialed digits. If the gateway gets an expected
sequence, the number will be accepted and sent for routing. Otherwise, the call will be refused. It is possible to
register plans counting on the dialing rules of your scenario and subsequently associate this plan to an E1 Link or
NAP FXS.

In order to validate the digits dialed, rules should be added to the dial plan settings. Each rule
determines the required prefix to validate the sequence.

Rules are determined using a sequence of values according to the syntax in the table below:

Digits 0123456789ABCD*#

Wild card X

Option [2578], [136], [29], ...

Interval [0-9], [3-6], [5-7], [0-2], ...

Option and interval [57-9], [0246-8], [2-579], ...

"X" can assume any numerical digit. I.e.; the rule "0XX4837222900" should match the sequence
"0214837222900" and fail with "0##4837222900".

Numerical digits inside "[]" mean a list of options and/or a numeric interval. I.e.; the rule "2[38][6-9]X"
should match the sequence "2370" and fail with "2470" or "2350". The wild card "X" corresponds to the "[0-9]"
interval.

Option and interval can be used in the same "[]" structure. I.e.; the rule "2[37-9][2-46]X" should match the
sequences "2360", "2830" and "2720". The same rule should fail with "2450", "2100" and "2070".

A sequence that partially matches a rule may be accepted in the dial plan. I.e.; the rule "XXXX"
Note accepts the sequence "123" or "456" after the signaling determines there are no more numbers
to be sent.

The sequence is validated per prefix. I.e.; the received sequence "12345" matches the "XXXX"
Note rule.

In the dial plan, rules are determined in a given order. The first rule has priority over the second, the
second rule has priority over the third, and so on. When a rule fails, it is removed from the checklist. When the
sequence matches, the highest priority rule in the checklist is applied. When the dialed digits match at least one rule,
the received sequence is accepted. When all rules fail, the call is refused.

7. SS7/SIGTRAN
"Configuration" → "Telephony" → "SS7/SIGTRAN"

This option is only enabled when KMG has SS7/SIGTRAN license.

Note
 Section containing the settings related to the use of SS7 and SIGTRAN interconnection protocols.

The KMG has support for SS7 and SIGTRAN signaling in its E1 modules. These signals are used for
inter-carrier trunking, and for interconnection with the core devices of their networks.

Khomp supports the MTP2, MTP3 M2PA and ISUP protocols. For more information on these protocols,
please refer to the ITU recommendations.

Special licenses are required to use SS7 or SIGTRAN signaling with the KMG. These licenses can be
obtained from Khomp's sales department. The licenses will be linked to the device's serial number.

7. SS7/SIGTRAN settings
Initially, you must configure all the links that are going to be used with ISUP signaling. Either the links
that will be used for signaling (MTP2 / M2PA) or the links that will be used for voice (Circuit Groups) or both.
Following that, go to the SS7/SIGTRAN menu on the KMG Web interface. To make configuration easier, have the
Interconnection Technical Project (PTI) at hand.

7. MTP2/M2PA
You must configure which of the links will be used as signaling links for the MTP2 and M2PA protocols.
There must be at least one link with signaling.

MTP2: Add the link defining its name, device, and mode of operation. Additionally, you can change the default
timers of the MTP2 protocol.

M2PA: Add the client or server M2PA link and define a name. Client: Enter the address and local port of the KMG, as
well as the address and remote port of the server. Server: Enter the remote client's address and port.

M2PA - Local port configuration: Local port used to receive the connection information, in case a server M2PA Link
is configured on this device.

7. Point Codes
Settings related to "Point Codes". There must be at least two configured Point Codes, local and remote.
When you fill in the decimal format, the 3-8-3 format field will be updated.

7. MTP3
Settings related to the MTP3 protocol. You must configure Linksets and Routes.

Linksets
Linkset settings for association with configured links. To add a Linkset, you must perform the following
procedures:

1. Select the Source (Source Point Code), which will be the point code used by this Signaling Point.
2. Select the Adjacent (Adjacent Point Code), which is the point code of the Signaling Point immediately
connected to this Linkset.
3. Select the Network Indicator.
4. Select which link will be used by this Linkset, defining the SLC (Signaling Link Code) value to be used by
this link (this value must be equal to the value at the adjacent signaling point).
Routines
Route configuration for MTP3.

To create a route, select the destination PointCode and associate the Linkset that can be used to reach
this point code.

If it is possible to reach this point code using more than one Linkset, click on "Add Linkset to Route" and
go back to step 3 as many times as necessary. Note that the order of inclusion will be the Linkset's priority order to
reach the configured point code.
 7. ISUP
ISUP circuit groups settings:

Circuit Groups
Circuit Groups settings and definition of timeslots to be used.
To add a Circuit Group:

1. Select the Source (Source Point Code), which will be the point code used as source of the calls.
2. Select Destination (Destination Point Code), which will be the point code used as destination of the
calls.
3. Select the link that will be used for voice traffic.
4. Define the initial Circuit Identification Code (CIC) value of this Circuit Group. This should be exactly the
same as the one configured at the other end of the link.

If there is any limitation regarding the link timeslots that may be used, check or uncheck the boxes
referring to the timeslots that should be used by this Circuit Group.
KMG does not allow the use of timeslot 0 and 16 as a circuit to be used in Circuit Groups
Attention configuration.

Timers
Definition of ISUP signaling timers.

7. Features

This menu presents all of the general resources for the equipment that may be acquired separately
depending upon the particular needs of each scenario.

7. Integrations
This menu presents all of the general resources for the equipment that may be acquired separately
depending upon the particular needs of each scenario.

7. Insight!
"Configuration" → "Resources" → "Integrations" → "Insight!"

Configurations to permit access through Insight! the Web interface of the KMG. To allow this, the VPN
configuration supplied by Insight must be loaded and the WEB portal configured for HTTPS.

7. Device Management
"Configuration" → "Resources" → "Integrations" → "Device Management"

Configurations for integration with the Broadsoft Device Management system that uses HTTP/HTTPS
requisitions to allow for the updating of the configuration and version of the KMG from a remote, centralized
location through the Broadsoft platform.

7. TR-069
"Configuration" → "Resources" → "Integrations" → "TR-069"

Configurations related to support for the TR-069 protocol allowing for updating of the configuration or
version of the KMG from a remote centralized location through an ACS server.
 7. Other systems
"Configuration" → "Resources" → "Integrations" → "Other systems"

To use this feature, you must have a GSM interface.

Note

Used for integration with the KSMS System which is Khomp’s platform designed to send SMS
messages, making it possible to reach clients through the use of SMS campaigns.

You will find more information about this platform at Khomp website. To use this feature, the KMG must
be connected to the Internet.

7. Survivability
"Configuration" → "Resources" → "Survivability"

This option will only be displayed when the KMG has a valid "Survivability" license.
Note

Survivability allows the KMG to be able to maintain the basic functionalities of a PBX in the event of
unavailability. The PBX is monitored and if a failure is detected the KMG can assume the role of the PBX and provide
resources in a limited manner, including the maintaining of registers and incoming and outgoing calls, using the
known registers and the routing table for the equipment for forwarding of calls.

The Survivability module is configured on the users side as a proxy for the PBX, so the entire call flow
goes through the KMG's Survivability module first, before it reaches the PBX. The Survivability module checks the
availability of the PBX at a configurable frequency (in seconds) via SIP OPTIONS. When there is no response to the
SIP OPTIONS command within the defined time frame, the KMG replaces it. At this point its operation mode
changes from proxy to Survivability mode.

We currently support the following operations in Survivability mode:

● Registration management.
● Local calls between extensions.
● Incoming calls (via E1/T1, GSM, or FXO KMG links)
● Outgoing calls (via E1/T1, GSM, or FXO KMG links)
● Call transfer (direct and assisted), both local and external

New extensions can register during Survivability mode, but need to register again once the PBX
becomes available.

All gateways of the KMG line support the Survivability module. Licenses must be purchased
Note separately.

Key
S = Survivability module
GW = Gateway
 7. Configuration
On the SIP NAP that will communicate with the desired PBX, configure the "Survivability" option in the
proxy list to enable this feature.

7. Network
"Configuration" → "Resources" → "Survivability" → "Network"

All KMG network interfaces will be available for configuration; select the desired protocol (UDP, TCP,
TLS), as well as the port through which the Survivability module should receive the packets.

The network interface address and the port configured for Survivability should be used as a
Note proxy setting for the PBX on the IP phone or softphone.

The following options are displayed:

Field Description Releas

e/Versi
on

Interface Identifies the network interface of the KMG's general network settings

IPv4 Settings to Use IP Version 4 4.0.5

IPv6 Settings to Use IP Version 6

IPv4 and IPv6 options

Field Description Releas

e/Versi
on

Name Network Interface name for configuring the Survivability mode

UDP Enables use of UDP transport on the selected port

TCP Enables use of TCP transport on the selected port 4.0.5

Enables use of TLS transport on the selected port

TLS
NOTE: Enabling TLS requires loading the keys into the KMG global TLS
settings.
 7. Servers
"Configuration" → "Resources" → "Survivability" → "Servers"

In this area you can configure the SIP server (PBX) that the Survivability module will monitor and take over in case of
failure.

When you click on "Add", a form will be displayed for registration of the server, containing the following
fields:

Field Description Release/

Version

Name Identification of the SIP server to be monitored. 4.0.5

Domain Identification of the domain that will be used for the user register
4.1.42.0
Address This is a list of addresses to be used for forwarding registers and calls.

SIP Port Network port used for communicating with the server.

Transport Transport used for communicating with the server 4.0.5

The network interface to be used for communicating with the SIP

Interface
server.
Keep Alive

Field Description Release/

Version

Time interval (in seconds) for the Survivability module to trigger tests
Interval between tests using SIP OPTIONS to detect failure/unavailability of the monitored
server.

Maximum number of Number of failures that must occur during the tests using SIP OPTIONS
failures before going into before the Survivability module takes over the communication from the 4.0.5
Survivability mode. monitored server.

Time (in seconds) for a registration to be considered expired when the

SIP registration expires in
Survivability module has taken over communication from the monitored
Survivability mode
server.

Field Description Release/

Version

Time interval (in seconds) for the Survivability module to trigger tests
Interval between tests using SIP OPTIONS to detect failure/unavailability of the monitored
server.

Maximum number of Number of failures that must occur during the tests using SIP OPTIONS
failures before going into before the Survivability module takes over the communication from the 4.0.5
Survivability mode. monitored server.

Time (in seconds) for a registration to be considered expired when the

SIP registration expires in
Survivability module has taken over communication from the monitored
Survivability mode
server.
 Calls

Field Description Release/

Version

Group of extensions for answering incoming calls; if not configured, all

registered extensions will be used as a group. The Survivability module
will attempt to complete incoming calls by trying one extension of the
group at a time, in the order defined here. Trunking extensions should
not be included in the answering group.
Extensions for answering
incoming calls
Note: When the Survivability module detects an incoming DDR call, the
DDR extension is entered as the first element of the group, and if it is in
the list, it may ring more than once.

4.0.5
Wait time (in seconds) for answering by one of the members of the
Time-out for call answering group. When this time limit is reached, the Survivability
answering module will cancel the call and make another call to the next extension
in the group.

Extensions (or registrations) that should be considered trunks by the

Survivability module. Extensions included in this list will be used to
make outgoing calls when in Survivability module; an outgoing call may
Trunking extensions attempt to use more than one trunking extension (in the order defined in
this setting). If none of the trunking extensions are available or this
setting is empty, the Survivability module will attempt to make outgoing
calls without registration to the routing module.

7. Register Authorization

This option is displayed only when KMG has a "Register Authorization".

Note

Configurations related to the Register Authorization module. This application allows external users to
access the PBX of the internal network using the KMG to authorize registers, providing security and topology hiding,
guaranteeing the identity and confidentiality of the users.

7. Network settings
"Configuration" → "Resources" → "Register Authorization" → "General" → "Network Configurations"

All of the network interfaces of the KMG are available for configuration. For each enabled interface,
inform whether the data transport will be UDP, TCP or TLS, and indicate the port that will be used. In the case of TLS,
choose the TLS setting in the corresponding field. To enable the TLS, access the menu "Configuration" → "System"
→ "VoIP" → "TLS Encryption".

The default value of the SIP protocol port is 5060

Note
 7. General options
● RTP ports: Enter the range of possible RTP ports for sending and receiving audio.
● Operation Mode: Select either "Forward" or "Back-to-back" mode, which will be explained further below.
Forward: In this mode, the registration requests received by KMG are forwarded to the first active server among the
servers set on the "Servers" screen.

Back-to-back: In this mode, registrations are handled independently for each configured server. To use this mode,
users must be pre-authenticated via query to an LDAP database. The LDAP database settings are as follows:

● LDAP Servers (Primary or Alternative): Configuring the LDAP database to check user registration
information.
○ Address: IP address of the server where the LDAP database is located.
○ Port: Port that will be used by the KMG to access the LDAP database. The default port for
LDAP is 389 and for LDAPS it is 636.
○ Schema: The type of database that will be used, which can be LDAP or LDAPS.

You must configure the Primary LDAP Server. Configuring the Alternative LDAP Server is optional. If configured, it
will be used if the query to the primary server fails.

● Perform BIND: If the LDAP database requests access credentials, the user and password can be
informed by selecting the Perform BIND checkbox.

● Search: Parameters for user search and evaluation in the LDAP server, through a rule that must be
entered and must be followed by users who try to register; depending on the result obtained with the
rule, an action can be taken.
○ DN Base: The verification rule that must be met for user validation, based on the ID and
values that will be extracted from the LDAP database.
○ Filter: Filter for comparison between the value extracted from the REGISTER message, and
the value extracted from the LDAP database, based on the DN Base rule. If the values are
identical, then the user will be allowed access.
○ Scope: Database level in which the value extracted for Register Authorization will be
searched in LDAP.
○ Password Attribute Name: Identifier used for the attribute used as password in the
database.

● Registration expiration: In the back-to-back mode, registration expiration times are independent for UAS
and UAC. Provide the expiration time for the UAS in the "Expiration time for UAS" field and the maximum,
minimum, and default values of the expiration time for UAC in the fields of the "Expiration Time for UAC"
section.

7. Advanced options
● Keep Encryption Key for Re-Invite: When using TLS, select this option to use the same encryption key
when a Re-Invite occurs.
● Strict address-of-record validation: It accepts subsequent messages after REGISTER (INVITES, etc.)
only if they are identified by the same address-of-record informed in REGISTER.
● Use the same UAC transport: Uses the same UAC transport for exchanging messages with UAS.
● UAC Keep Alive (OPTIONS): Sends Keep Alive, via OPTIONS packages, to UAC. For each type of
transport (UDP, TCP or TLS), select the interval in seconds to send the packages, or use the "Disabled"
option, so as not to send Keep Alive.

Click on "Save" to store the settings, then click on "Apply" to send them to KMG
Note
 7. Media profiles for Register Authorization
Media profiles save the Codecs that can be used by users of the Register Authorization service when
making calls through the KMG. They can be configured at:

"Configuration" → "Resources" → "Register Authorization" → "Media Profiles"

To add a new profile, click on the Add button. In the screen that will be displayed, name the profile, select
the allowed Codecs, and the media transports (UDP-RTP, UDP-SRTP or TCP-RTP) related to each signaling transport
(SIP UDP, SIP TCP or SIP TLS ).

Click on "Save" to store the settings, then click on "Apply" to send them to KMG
Note

7. Servers for Register Authorization

KMG authorizes new registrations or not; when authorized, it sends the registration to the configured
servers according to the selected operation mode (Forward or Back-to-back). To configure the servers, go to:

"Configuration" → "Resources" → "Register Authorization" → "Servers"

To add a new server, click on the Add button and configure the following parameters:

● Name: Server identification name.

● Domain: Domain associated with the server.

In the Addresses panel, configure the following server-related fields:

● Address: IP address of the server.

● Port: Network port that will be used to access the server.
● Transport: Type of data transport between the KMG and the server, with the options of UDP, TCP or TLS.
● Keep Alive (OPTIONS): Time interval for sending Keep Alive, a message that verifies that the network
structure is connected and working.
● Network Interface: KMG network interface that will be used to communicate with the VoIP server;

Each server can have more than one address. To add a new address, click the Add Endpoint link.

● Operation mode:
○ BRIDGE: There is no handling of the call media by KMG, therefore, there is no transcoding.
Select a media profile for UAC and one for UAS. KMG will accept calls with any Codecs, even
if different from those configured in the media profile.
○ TRANSCODE: It allows conversion between different Codecs. Select a media profile for UAC
and one for UAS.
○ FILTERED-BRIDGE: Call forwarding ONLY when the telephone device or server supports the
same Codecs that are available for use in the KMG. In this case there is no transcoding, and
if the telephone device or the server does not display a compatible Codec, the call is
ignored. The accepted Codecs will be those configured by the chosen media profile.

Click on "Save" to store the settings, then click on "Apply" to send them to KMG
Note
Configured servers will be shown in a list. To make a server the primary one for sending messages,
press the icon indicated by the star (Set UAS as primary).
 7. Policies for Register Authorization
With the Register Authorization policies, you can allow or not users to register and make calls through
the KMG. Verification is performed through the analysis of SIP protocol methods, such as REGISTER, INVITE, etc.
To configure the policies for Register Authorization, go to:

"Configuration" → "Resources" → "Register Authorization" → "Policies"

Policies can be used together, and a register will only be allowed if it meets all the rules contained
therein

7. All methods: User-Agent validation

By means of regular expressions, it allows, blocks or forwards the verification to the other methods, based on the
received User-Agent header. If the message does not contain the expected field, it is considered empty (regular
expression ^$).

● Regular Expressions for User-Agent Validation: Regular expressions that must be checked so that
actions can be taken.
● Action if evaluated positively: If the value extracted from the User-Agent header meets the established
rule, the configured action is taken.
● Action if evaluated negatively: If the value extracted from the User-Agent header does not meet the
established rule, the configured action is taken.

Possible actions are:

● ACCEPT - Accept the request.

● DROP - Ignore the request.
● NEXT - Forward the request to the next policy level.

7. REGISTER Method: pre-authorization

Verification done when a telephone set tries to register to the IP PBX through the KMG.

Below, you can see an example of a REGISTER type message:

REGISTER sip:10.3.0.34;transport=TCP SIP/2.0

From: "Khomp
00"<sip:[email protected]>;tag=a92908-0-13c4-58ac54c5-60440b51-58ac54c5
To: "Khomp 00" <sip:[email protected]>
Call-ID: aa44d0-0-13c4-58ac523f-1df7d2a3-58ac523f
CSeq: 3 REGISTER
Via: SIP/2.0/TCP 10.100.10.246:5060;branch=z9hG4bK-58ac54c5-612b2463-2af4becf
Expires: 3650
Allow: INVITE, CANCEL, BYE, REFER, NOTIFY, SUBSCRIBE, INFO, ACK, MESSAGE
user-Agent: IPS 200 V4.0.2.6 9726
Max-Forwards: 70
Supported: replaces,100rel,eventlist
Contact: <sip:[email protected]:5060;transport=TCP>
Authorization: Digest
username="3321000",realm="asterisk",nonce="1487699055/ebd4bf6fd11830a630fe61e
9587def33",uri="sip:10.3.0.34;transport=TCP",response="d9f680586ffb746cae9dfbfa0ff0
46ab",
algorithm=MD5,cnonce="61
2464",opaque="28a16df8265154d3",qop=auth,nc=00000002
Content-Length: 0
 ● Use SIP header values - The following SIP message fields can be used for verification:
○ AUTHORIZATION - User authentication data.
○ FROM - Message origin identification.
○ TO - Message destination identification.

● LDAP Configuration - Configuration of the LDAP database to perform the verification of extracted
information in Use SIP header values:
○ Address: IP address of the server where the LDAP database is located.
○ Port: Port that will be used by the KMG to access the LDAP database. The default port for
LDAP is 389 and for LDAPS it is 636.
○ Schema: The type of database that will be used, which can be LDAP or LDAPS.

● Perform BIND - If the LDAP database requests access credentials, the user and password can be
informed by selecting the "Perform BIND" checkbox.
● Search - Parameters for user search and evaluation in the LDAP server, through a rule that must be
followed by users who try to register. Depending on the result obtained with the rule, an action can be
taken.
○ DN Base: The verification rule that must be met for user validation, based on the ID and
values that will be extracted from the LDAP database.
○ Filter: Filter for comparison between the value extracted from the REGISTER message, and
the value extracted from the LDAP database, based on the DN Base rule. If the values are
identical, then the user will be allowed access.
○ Scope: Database level in which the value extracted will be searched in LDAP database.

● Attribute Validation - Action that must be taken according to the result obtained in the evaluation of the
attempt to register.
○ Action if evaluated positively - If the value extracted from the message meets the
established rule, one of the following actions can be taken:
■ ACCEPT - Accept registration.
■ DROP - Ignore registration.
○ Action if evaluated negatively - If the value extracted from the message meets the
established rule, one of the following actions can be taken:
■ ACCEPT - Accept registration.
■ DROP - Ignore registration.
○ Action if it is not possible to evaluate - If the value extracted from the message meets the
established rule, one of the following actions can be taken:
■ ACCEPT - Accept registration.
■ DROP - Ignore registration.

7. INVITE method: Checking the registration status

Verification of a user register status, made when they try to make a call (INVITE) through the KMG. Example of
INVITE type message:
 INVITE sip:[email protected];transport=TCP SIP/2.0
From: "Khomp
00"<sip:[email protected]>;tag=a86188-0-13c4-58ac5814-13bba396-58ac5814
To: "3321002" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/TCP 10.100.10.246:5060;branch=z9hG4bK-58ac5814-61380f03-7086c167
user-Agent: IPS 200 V4.0.2.6 9726
Max-Forwards: 70
Supported: replaces,100rel,eventlist
Allow: INVITE,CANCEL,BYE,REFER,NOTIFY,SUBSCRIBE,INFO,ACK,UPDATE,MESSAGE
Contact: <sip:[email protected]:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 251

v=0
o=IpPhone 2890844526 8000 IN IP4 10.100.10.246
s=IpPhone CALL
c=IN IP4 10.100.10.246
t=0 0
m=audio 10006 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

● Use SIP header values: The following SIP message fields can be used for verification:
○ AUTHORIZATION - User authentication data.
○ FROM - Message origin identification.
○ TO - Message destination identification.

● Positive if status is one of the following: The evaluation is considered positive if the register status is:
○ INVALID - The status cannot be checked.
○ REGISTERED - The user is registered.
○ EXPIRED - The registration session has expired.

● Action if evaluated positively:

○ ACCEPT - Accept the message.
○ DROP - Ignore the message.

● Action if evaluated negatively:

○ ACCEPT - Accept the message.
○ DROP - Ignore the message.

● Action in case it is not possible to evaluate:

○ ACCEPT - Accept the message.
○ DROP - Ignore the message.

Click on "Save" to store the settings, then click on "Apply" to send them to KMG
Note
 7. Global settings for Analytics
"Configuration" → "Resources" → "Analytics"

This option contains the global KMG Analytics timing settings. The values provided in this menu affect
the audio frequency analysis performed by Analytics.

Those settings usually cover most scenarios in telephony. Change these properties only if
Attention you are knowledgeable or have orientation from Khomp support.

Field Description

Necessary duration of silence, after a voice is detected post-answering, to

Response silence duration
report that a human answered.

Silence time Silence time post-answering, so that silence is reported.

Necessary duration of silence, after a voice is detected post-answering, to

Response silence duration
report that a human answered.

Maximum duration to detect that a Maximum duration of the call during detection, done after Response
human answered silence duration, in order to consider that a human answered.

Maximum duration for detection of Maximum voice time post-answering, to consider that it was a short
short human answer human answer.

Minimum voice time to report Minimum voice duration before connection to report Unknown Message
Unknown Message during answering.

Minimum time to keep ringing so that interception signaling is not

Minimum time with ringing tone
detected.

7. Routing
The routing menu brings together all of the configurations related to call routing, which are the most
common configurations of the equipment.

In the Routing menu, you can configure the routes and NAPS that connect calls from one point to
another. In addition, profiles, behavior of Analytics (if the license is applied), CDR, Portability, and Routing scripts are
defined, among other functionalities.
 7. NAP
"Configuration" → "Routing" → "NAPs"
As mentioned previously, the KMG functions according to the concept of creating associations between
NAPs and Routes. NAP - Network Access Points are logical representations to group E1/T1 links, GSM, channels
analog FXS and FXO channels, and SIP trunks. These NAPs are used for routing calls involving points of origin,
destination, or a combination of the two. Therefore, before configuring a route, it is necessary to determine its
incoming and outgoing NAP.
When selecting the type of NAP, the fields for configuration will be displayed, some of which are
exclusive and correspond to the technology selected. More details about each type of NAP are described
throughout this section.

7. Creating an E1/T1 trunk

"Configuration" → "Routing" → "NAPs"

1. Click on the "New NAP" button.

2. In the "Type" field, select the "Trunk" option. Options for configuring an E1/T1 trunk will be displayed.

3. Select a telephony profile, or leave the option "Use default profile", which serves the majority of
scenarios. See the section "Configure call properties" for more information on profiles.
4. In the "Script for Routing" field, select a previously created script. Please refer to the "Scripts" section, in
this manual, for more information.
5. In the field "Channel allocation algorithm", select an algorithm to define the way in which the link’s
channels will be used. Please refer to "Channel allocation algorithm", in this manual, for more
information.
6. The panel "Devices" will display all of the modules connected to the KMG that have an E1/T1 interface,
represented by their serial numbers.
○ Unallocated: The link will not be used by this NAP.
○ The entire link: All of the channels of this link should be used by this NAP. If this option is not
being displayed, the link is being used in a fractioned manner by another NAP.
○ Fractionate Link: Only the selected channels will be used by this NAP. If a channel is
disabled, it is being used by another NAP.
○ If you are creating an E1 trunk and you see only the message "This link is already completely
allocated by another NAP", this means that all of the E1/T1 links available are being used by
another NAP.
 7. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes to
take effect.
Devices that have E1/T1 links that are fully allocated will display a message stating this situation. During
routing, this NAP is valid as source if the channel allocated for incoming call is associated with it.

7. Creating an SIP trunk

An SIP NAP is an entry/exit point for VoIP calls. When selecting this type of NAP, the following fields are
displayed.

Field Description Release/

Version

Domain Host IP address.

-
SIP domain listening port, where the packets will come from and where
Domain Port
they should be sent to.
Registration

Field Description Release/

Version

Check this box if the domain requires registration and authentication to

Registration
send/receive calls.

User Registered domain user name.

Authorization User Username that must be used for authentication -

Password Password for authentication at the domain.

Time interval in seconds required for SIP registration expiration.

Expiration
(Default time 3600 sec.)

Default value: Always. In this case no analysis will be performed and it

will accept the NAP as valid. The other behaviors refer to the fields
Accept this NAP as
received in the Invite. A verification is performed to check whether the 3.3.0
source
registered user is included in the From, To, or both fields; the incoming
call is accepted or not depending on this analysis.
Options

Field Description Release/

Version

Use IP packet address as In incoming calls, use the IP packet address as the "Domain" for
source validation, rather than using the value contained in the "From" field.

Force incoming calls

Only calls originating from the proxy servers defined in the "Proxy List"
through a configured
will be accepted
proxy

Ignore source port In incoming calls, it ignores the SIP packet source port.
-

Use called number from In incoming calls, it uses the called number sent in the "RequestURI"
URI header instead of using the value in the "To" field.

The time limit in seconds for the proxy configured in this NAP to be considered as a
destination. If no answer is received within this time limit, the next proxy on the list
Proxy fallback timeout will be used. If all of the destinations are unavailable, this NAP will not be used and
a retry will be performed using a different route in accordance with the
configuration.
 Field Description Release/
Version

Time-out for idle SIP

Time interval to terminate an idle SIP call. -
session

Time interval between Enables or disables Keep Alive checking of the remote SIP server. If
Keep Alive Checkups (SIP enabled, this option will disable NAP if the Keep Alive check fails. 3.2.0
OPTION) Defined time in milliseconds.

Limit of allocated Limits the number of VoIP channels that can be used by this NAP. This
3.0.0
channels limit applies to incoming and outgoing calls.

Network

Field Description Release/

Version

Network Interface for SIP

Defines the network interface that should be used for the SIP protocol.
Protocol

Network Interface for

Defines the network interface that should be used for the RTP protocol. 3.2.0
RTP Protocol

NAT traversal mode Enables or disables NAT traversal support.

If NAT traversal by "STUN Server" is enabled, this field must be filled in

STUN
with the STUN server IP address.
2.1.0
If NAT traversal by "STUN Server" is enabled, this field must be filled in
STUN port
with the STUN server network port.

External IP address for If NAT traversal by "Fixed IP external address" is enabled, this field must
the SIP protocol be filled in with the external IP address used for the SIP protocol.
3.2.0
External IP address for |If NAT traversal by "Fixed IP external address" is enabled, this field
the RTP protocol must be filled in with the external IP address used for the SIP protocol.

Transport

Field Description Release/

Version

Accept any type of SIP

transport in incoming Accepts any incoming SIP signaling protocol for this NAP.
calls
4.0.0
Accept any type of audio
transport (RTP) in Accepts any incoming RTP audio protocol for this NAP.
incoming calls

Type of SIP transport Defines the type of transport used for the SIP protocol.
3.1.0
Type of audio transport Defines the type of transport used for the RTP protocol.
Proxy list

Allows you to add SIP proxy servers. To add a new server, click on "Add proxy" in the lower-right corner
of the proxy list. In the SIP Interface, SIP Transport, Audio Interface and Audio Transport options, you can use the
settings applied to SIP NAP, or specify different values.

7. Configuring GSM channels

GSM NAP creates a point for incoming and outgoing calls through this interface.

7. Options
The field "Channel allocation algorithm" determines which algorithm should be used to allocate the
channels associated with this NAP. Please refer to "Channel Allocation Algorithm" section for more information.

The "SMS Channel Allocation Algorithm" field allows you to configure the use of SIM cards to send SMS,
in which the usage algorithm and the sorting are defined.

● Algorithm
○ First free: Allocates the first available channel.
○ Round robin: Allocates channels in a balanced way so that all of them send the same
amount of SMS

● Sorting channels:
○ Ascending: Allocates from the lowest channel to the highest channel.
○ Descending: Allocates from the highest channel to the lowest channel.

SIM cards
Provides options for using SIM cards in each GSM device modem. It is worth noting that each modem
has an active and a stand-by SIM card, and they can be switched.

● Minutes of use to change SIM card: When this option is enabled, you must specify the maximum
number of minutes, according to the agreement with the carrier. When the maximum number of
minutes is reached, KMG forwards the call to the stand-by SIM card.
● SMS messages sent to change SIM card: When this option is enabled, you need to specify the
maximum number of SMS messages that can be sent by the SIM card, according to agreement with the
carrier. When the set number of SMS messages is reached, KMG will forward messages to the stand-by
SIM card.

By clicking on the "Add manually" button, you can enter the SIM cards. To do this, the SIM card ICCID
must be filled in.

Usage control
If this option is enabled, it provides an option to control the use of SIM cards individually, regardless of
the model in which the SIM card is inserted.

● Period for renewal: Determines the period after which the SIM card can be used again, if the limit of
minutes or SMS messages has been reached. You can select the following values: Daily, Weekly or
Monthly. Depending on the selected period, you need to specify the renewal time in the following field.
For example, if the selected period is Daily, you must specify the renewal time. If you select Weekly, you
must specify the day of the week, and the day of the month for the Monthly period.
● Minutes per period: Number of minutes that a SIM card can use during the period set. When this limit is
reached, another SIM card will be used.
● SMS messages per period: Number of SMS messages that a SIM card can send during the period set.
When this limit is reached, another SIM card will be used.
● Shared limits: If this option is enabled, both the number of minutes and SMS messages will be shared
among all SIM cards available in the NAP.
● Interval for usage control: Determines the time interval for controlling usage of the SIM card plan. The
first value indicates the initial billing time, regardless of the call duration. The second value indicates the
time for charging additional minutes. So if 60/6 is set, the call will be charged for 60 minutes, even if the
call duration is less than this time. If this same call is longer than this time, the billing control will be
every 6 minutes.
● Change SIM Card: Determines the time when the SIM card should be changed.
 7. Configuring FXS channels
The FXS routes are the associations between the KMG and a destination FXS NAP, which will be used to
allocate the FXS channel for sending. You can also define filters for its use, such as the destination number and a
portability query.
Create a new NAP and select the type as FXS; note that an additional menu will be displayed with the
following options:

● End-of-dial marker - A key that when dialed will mark the end of the dialing process.
● Time-out for dialing - time-out to dial the call; after the time limit expires, the call will be canceled.

Still in the additional menu, select one or more channels to compose the NAP and make the following
configurations:

7. Registering extensions
Field Description

Domain Host IP address

Domain listening port, where the packets will come from and where they should
Domain Port
be sent to.

Expiration Time for expiration of the validity of the registration.

Options

Field Description

Use IP packet address as In incoming calls, use the IP packet address as the "Domain" for validation, rather
source than using the value contained in the "From" field.

Maximum wait time for a response via a given active proxy. Upon reaching
Force incoming calls through time-out, the next active proxy will be used (fallback). When there are no active
a configured proxy proxies to perform fallback, either the call is terminated or there is a retry on the
next route.

Ignore source port In incoming calls, it ignores the SIP packet source port.

In incoming calls, it uses the called number sent in the "RequestURI" header
Use number called in the URI
instead of using the value in the "To" field.

Maximum wait time for a response via a given active proxy. Upon reaching
time-out, the next active proxy will be used (fallback). When there are no active
Fallback time-out
proxies to perform fallback, either the call is terminated or there is a retry on the
next route.

Time interval between Keep

Enables or disables Keep Alive checking of the remote SIP server. If enabled, this
Alive Checkups (SIP
option will disable NAP if the Keep Alive check fails. Defined time in milliseconds.
OPTION)

Time-out for idle SIP session Time interval to terminate an idle SIP call.

Enables or disables the use of certain Codecs according to the created CODEC
Route CODEC profile
profiles.
 Network

Field Description

Network Interface for SIP

Defines the network interface that should be used for the SIP protocol.
Protocol

Network Interface for RTP

Defines the network interface that should be used for the RTP protocol.
Protocol

NAT traversal mode Enables or disables NAT traversal support.

Transport

Field Description

Accept any type of SIP Allows for receiving calls regardless of the data transport protocol used at the
transport in incoming calls source.

Accept any type of audio

Allows for receiving calls regardless of the audio transport protocol used at the
transport (RTP) in incoming
source.
calls

Type of SIP
Selects the data transport protocol of the calls.
transport

Type of SIP
Selects the audio transport protocol of the calls.
Audio

7. Proxy list
If the IP PBX server has proxies, they must be entered and configured in the Proxy List. Indicating the
network address, and the port by which you can access the server and transport the KMG data.

7. Extensions

Table in which the extensions of the FXS devices are registered. This information must match those of
the SIP server.
In the FXS device table header, the serial number is displayed. Beside it, there is an option to fill in the
extension number and user for automatic registration. For this, you only need to register the lowest extension in the
range.
For example: When you register extension 2000 through this option, the KMG automatically registers the
other extensions (2001, 2002, etc.) for all channels of the device.
Lastly, there is a button to clear the extensions that have been registered on the device, if it is necessary
to clear them all.

Below is a description of each column in this table.

● Channels: List of FXS channels that can be included in the NAP (channels with a blocked selection are
already allocated to an FXS NAP).
● Extension Number: Dialing number that identifies the channel;
● User for registration: User for extension registration;
● Password for registration: Password for extension registration.
 7. Ring pattern
Defines the ring pattern for the devices defined in the NAP, in milliseconds. You must register at least a
ring duration and a silence duration.

By default, the KMG has a ring duration of 1000 milliseconds and 4 milliseconds of silence.

7. Cadences
Sets the duration of the dial tone and disconnect tone that will be transmitted to the devices registered
in the NAP. You must register at least a tone duration and a silence duration.

7. Configuring FXO channels

FXO routes are the associations between the KMG and a destination FXO NAP, which will be used to
allocate the FXO channel for receiving calls.

Field Description

List of FXO channels that can be included in the NAP (channels with a blocked
Channels
selection are already allocated to a FXO NAP).

Number of this line Number that will identify the channel.

7. Configuring NAP groups - NAP GROUP

NAPs GROUP represent a grouping of other NAP types, each associated with a weight. When a NAP of
this type is associated to a route as outgoing NAP, outgoing calls will be balanced between the members of the
group, according to the selected distribution algorithm. When associated to a route as an incoming NAP, all
members of the group are valid candidates as the entry NAP for that route.
Groups never have an associated profile; forwarded calls must use the route profile of the member or
default NAP.
The "dest_nap" key in the CDR will contain the name of the member NAP used to forward the call and
the "nap_group" key should be used to register the group name.

Field Description

When this option is enabled, the gateway will disregard group members that have
exceeded their call forwarding weight (refer to "Weight"), even if there is a retry
Force proportional
going on within the group). If it is not possible to forward the call to any of the
distribution of weights
members (because the proportional value has been exceeded or because there is
no available channel), then the system should consider the next route

Selects the algorithm used to perform the load distribution between the group
Load distribution algorithm
member NAPs.

NAP Selects the name of the group member NAP

The weight assigned to each NAP will be normalized using the following formula:
100 * WEIGHT / SUM(WEIGHT), providing a number within the range [0–100],
Weight named proportional weight, which will be used for channel allocation distribution
within the group. Use the zero value for weight for the NAP to be considered for
allocation only during a retry within the group
 7. Load balancing algorithms
Two algorithms are available: "Forwarded Calls" and "Active Calls".

● Forwarded calls: The number of calls that were proportionally forwarded to each group member NAP
will be considered. This algorithm aims to forward calls proportionally in relation to the assigned
weights.
● Active Calls: The number of active calls at the time of forwarding for each member NAP in relation to
the total active calls for all members will be considered. This algorithm aims to keep active calls
proportionally in relation to the assigned weights.

The NAP whose calculated index is farthest from its proportional weight will have the highest priority.
Members that have an inactive status or that have all channels occupied, will be disregarded.
In the case of "retry" within the group or "fallback", the proportionality of the weights can be ignored,
allowing calls to be forwarded to members that have a resulting index greater than the proportional weight, when
the option "Force proportional distribution of weights" is disabled.
Group members with a zero weight value will be considered only in the "retry" within the group or
"fallback" situation. They are not considered for generation of the indexes and never take precedence over the
members that have assigned weights.

When you change and reload the configuration of a group, the call forwarding
Note counters will be reset.
Retry within the Group and Fallback

If a call is forwarded to a member of the group and this member is not able to complete the call, then
there may be a retry within the group. In this situation, the next member will be used to perform a new call
forwarding. This new forwarding is considered for the use of the Forwarded Calls algorithm. The occurrence of a
retry within the group follows the same rules as of the "retry" for the system routes, so if the disconnection cause is
not enough to justify a "retry", it will not occur within the group either.
If a call is forwarded to a member of the group that supports fallback and it occurs, then the same
member of the group will be used to forward the call again in fallback mode. This new forwarding is considered for
the use of the Forwarded Calls algorithm.
In both cases, if the option Force proportional distribution of weights is enabled, it may consider using
the next gateway route instead of performing a retry within the group or fallback.

7. Channel allocation algorithm

In each NAP it is possible to define the configurations for allocation of channels. There are two settings;
the first one is the algorithm to be used, and the second one is the algorithm configuration, whenever available.
Two allocation algorithms are available: First Free and Round-robin. They may be configured for
different forms of allocation, according to the NAP type.
It also offers the possibility of making the allocation configuration automatic, which is the default value
for all NAP types.
 7. Channel allocation: automatic mode
Allows the KMG to automatically select the algorithm to be used. Applying this setting may cause
behavior to vary between versions.
Currently the automatic behavior uses the algorithm "First Free", configured using "Channels Sorting" in
"Ascending mode".

7. Channel allocation: first free channel

This algorithm will always test channels starting from the first available to the last, in search of an
available channel. The first available means the first channel resulting from the sorting configuration.
You can sort the channels by their index, using "Channel Sorting" or, if the NAP is of the "Trunk" type, you
can use "Sorting by Links", in which it will consider the channel index and the link to which it belongs.
The use of this algorithm is recommended whenever the carrier uses a similar allocation algorithm for
incoming calls, with the objective of avoiding competition in channel allocation between the KMG and the carrier.
For example: The carrier and the PBX always allocate from the lowest to the highest channel available,
thus, the KMG can be configured to always allocate inversely, i.e. from the highest to the lowest channel available.

7. Round-robin (circular allocation)

This algorithm will always test channels starting from the next channel relative to the last allocated
channel, in search of an available channel. By "next", we are referring to the next channel relative to the last one, i.e.
the next channel resulting from the sorting configuration.
You can sort the channels by their index, using channel sorting or, if the NAP is of the Trunk type
(E1/T1), you can use "Sorting by Links", in which it will consider the channel index and the link to which it belongs.
We recommended the use of this algorithm for GSM NAP and whenever the largest call flow is outgoing
to a NAP Trunk.

7. Sorting options
Channel sorting will sort the channels considering their index and the telephony module. They will be
sorted according to the selected option which can be "Ascending" or "Descending".
For devices with multiple telephony modules, channels with similar indexes, but located in different
modules, will be grouped and sorted using the serial number of the device as a reference.
Sorting by links will sort the channels considering their index and the telephony module. Channels will
be sorted according to the selected option, which may be "Lowest channels first" or "Highest channels first".

For example: For a Trunk-type NAP configured with two links from the same telephony module, say link
1 and link 3, opting for the Round-robin algorithm and ordering by Links configuration, using the "Lowest Channels
First", the behavior for four consecutive outgoing calls will be:

1. Call 1 will use the channel (timeslot) 0 of link 1.

2. Call 2 will use channel 0 of link 3.
3. Call 3 will use channel 1 of link 1.
4. Call 4 will use channel 1 of link 3.

7. Routes
"Configuration" → "Routing" → "Routes"

A table is displayed showing the configured routes. To create a new route, click on the "New Route"
button. The "Options" column displays the icons for editing or deleting a route.
 Field Description

Name Name for route identification

Priority to check a route over the others, if the route is similar to another. This
Priority value varies between 0 and 99; 0 indicates maximum priority, while 99 indicates
minimum priority.

Source NAP Select the source NAP for the call.

The called number. The call data must match those in this field for this rule to
B Number
apply. Accepts regular expressions (POSIX.1-2001).

The caller number. The call data must match those in this field for this rule to
A Number
apply. Accepts regular expressions (POSIX.1-2001).

This option is displayed only if portability is enabled. Please refer to the

Portability section in this manual for more information. If enabled, it checks
Check Portability
whether the number to be dialed belongs to the carriers specified in the "Valid
service providers" field.

This option is only displayed if the "Check portability" field is enabled. Selects the
Valid service providers
service providers that are valid for this route.

Change number for It is used to format the number to be checked, according to the format expected
portability query by the query service provider. Accepts regular expressions (POSIX.1-2001).

Destination NAP Select the NAP to which the call will be forwarded if all rules are met.

Changes the B number (called) of the outgoing call by using regular expressions
Change B number
(POSIX.1-2001).

Changes the A number (caller) of the outgoing call by using regular expressions
Change A Number
(POSIX.1-2001).

Selects the Profile used by the destination. Profiles are defined in the Profiles
Destination profile
menu.

Select the predefined restriction in "Time Restriction". This rule will only apply on
Time Restriction
the dates and times specified in the restriction.

This option is only displayed if the source NAP is SIP or E1, with R2 or ISDN
signaling. Filters at the source channel according to the selected option. For SIP
NAPs, it is necessary to indicate the header of the collect call in the Profile.
Collect call filter
● Accept all calls: All calls are accepted.
● Accept collect calls only: Only calls signaled as collect are accepted.
● Reject collect calls: Calls signaled as collect are not accepted

This option is only displayed if the destination NAP is VoIP or E1, with R2 or ISDN
signaling. Signal outgoing calls according to the option selected. For SIP NAPs, it
is necessary to indicate the header of the collect call in the Profile.
Collect call indicator
● Paid calls: Calls are signaled as paid.
● Collect calls: Calls are signaled as collect.
● Forward signaling: Calls are signaled according to the incoming call
 7. Regular expressions
Regular expressions follow the extended POSIX standard. Its official definition is available in chapter 9,
session 4 of the POSIX specifications. For more information, use the full reference (in English).

Examples of regular expressions

^[0-9]{10}$ → Filter, accepts exactly any 10 digits.

^10385555[0-9]{8}$ → Filter, accepts the prefix 10385555 + any 8 digits
/(.{10})/021$1/ → Replacement, forwards 021 + first 10 digits
/^10385555([0-9]{8})$/$1/ → Replacement, forwards only the last 8 digits (removes the
prefix)

Example of a configured route

Name LOCAL_NUMBER

PRIORITY 1

SOURCE NAP SIP_DIALER

B NUMBER ^48[0-9]{8}$ Only accepts numbers with the prefix 48 + 8 digits

A NUMBER Blank (does not distinguish by A number)

DESTINATION NAP CARRIER_01

CHANGE B NUMBER /^48([0-9]{8})$/$1/ Removes the prefix and dials only the last 8 digits

CHANGE A NUMBER Blank (does not change the A number)

7. Retry
By default, when the defined route for call forwarding defines an unavailable NAP (physical or logical
failure) as a Destination NAP, that route is ignored and the next valid route is used.
If, after the call is routed, there is a call failure, e.g., a congested network, the call can be forwarded using
the next valid route, as long as the failure cause is marked for Retry in the settings profile of the source call.
Routes whose filters accept the source call are considered a valid route. If there is no valid new route,
the call will be rejected.

Reloading the gateway settings will disable the "retry" feature for all ongoing calls.
Note

7. Configuring call profiles

"Configuration" → "Routing" → "Profiles" → "General"
 Profiles define behavioral patterns for the channels during calls. During a call, it always uses two
profiles, one for the channel associated as a source and another for the channel associated as a destination, which
may be different. Some profile properties are valid for both associations, others are valid only for source and others
only for destination. As an example, a cadence generation setting is valid only when the channel is associated with a
call as a source.

This option displays the profiles registered in the KMG for use in NAPs and routes. In the option column,
there is a button to make this profile the system default, which is represented by a star. You can also copy an
existent profile by clicking on the button represented by a double sheet.

To add a new profile, click on the "Add" button.

Field Description NAP available Version

for included
association

Name Name to be assigned to this profile. - -

Disables sorting or defines which set of Analytics

Call classification Destination 3.0.0
rules should be used.
Audio

Field Description NAP available Version

for included
association

Echo Cancellation Enables or disables echo cancellation.

-
DTMF
Enables or disables DTMF suppression.
suppression Any

Disables, changes with fixed gain or enables automatic

volume control. When defining manual control, the
Volume control 2.1.0
volume ranges between -10 and 10, where each unit
represents 3d.
Fax

Field Description NAP available Version

for included
association

Defines which behavior to adopt when a FAX signal is

Fax support detected. The behavior adopted will vary depending on Any -
the signaling.
 General

Field Description NAP available Version

for included
association

Enables the retry option and allows you to set up the

maximum wait time for the allocation of the outgoing
"Retry" support
channel, as well as the causes that will be used in the
call records.

Maximum wait
time for the
Maximum wait time before the KMG uses another valid
outgoing channel
route.
allocation to
perform "retry"

Selects which preferred disconnection cause will be Any -

used to map the cause of disconnection for the other
channel participating in the call, in case the system has
more than one disconnection cause available to map
and the call has not been completed.
Preferred
For example: If the gateway has performed more than
disconnection
one "retry" and the call could not be completed, it is
cause for
possible to use the cause in which there is no available
mapping
route, i.e., the gateway cause. Use either the cause
reported by the First disconnected channel, the one that
originated the first "retry", or the cause of the Last
disconnected channel, whichever one tried to complete
the call last.

Enables or disables display name forwarding in VoIP or

Forward Display
ISDN destination calls. For ISDN the value is limited to 80 Destination 4.0.0
Name
ASCII characters.

Allows for enabling custom cadences and configuring

Cadences Any -
them.

R2

Field Description NAP available Version

for included
association

Time-out for Maximum wait time for reconnection during a double

reconnection answering.

Time to wait for

Maximum wait time for receipt of the cause attributed to
condition of the
the incoming call before sending the default condition
incoming call
Destination -
Default condition
Default condition to be sent if the wait time for the
of the receiver of
condition of the incoming call is reached
the call

Forced
Enable sending of forced disconnection in the R2
disconnection
 SIP

Field Description NAP available Version

for included
association

Option that makes it possible to configure the type of

DTMF sending default for DTMF to be used for sending. The options
Any 4.0.0
type available are In Band, Out Band RTP (RFC 2833) and
three types of Out Band Sip Info

Send Custom Option in which you can enable the use of a custom SIP
Any 2.1.0
Header header

Identification
Option that makes it possible to configure the collect call
heading for
identification heading at the origin and the value that will Source 4.1.43.0
collect call at the
be used to send or receive collect calls a in the SIP
origin

Collect call Option that makes it possible to configure the collect call
identification identification heading at the destination and the value Source
4.1.43.0
heading at the that will be used to send or receive collect calls in the
destination SIP

Forward
Enables or disables extension header forwarding for
extension headers Any -
VoIP-VoIP calls.
(X-Headers)

Forward
The gateway must forward the header or User-to-User
User-to-User Any -
information from the incoming call to the outgoing call.
information

Enables or disables immediate audio opening. If enabled,

it will be possible to generate the "Connecting" cadence
on SIP source channels. If disabled, the audio opening
Enable early Source
will occur when there is confirmation of destination 3.3.0
media behavior
channel allocation; the "Calling" cadence will be
generated or the destination channel audio will be
replaced, if any.

SIP: Enable Enables sending of the Contact header in the SIP

sending of the message 180 Ringing. If this option is disabled, the Source
4.0.0
Contact header in header will be added when the NAT traversal mode
the 180 Ringing option of the NAP is configured.

Use the reason

field in the This configuration makes it so that the SIP cause
Source
heading as the (status-code) is not considered whenever the field 4.0.0
cause for Reason is available for mapping of the cause
disconnection

Enable tone detection for FAX and DTMF in VoIP

Activate tone channels. When enabling this option, there will be higher Source
4.1.42.0
detection equipment processing consumption, and the maximum
number of simultaneous calls will be affected

There is a maximum limit of 5 SIP headers to be forwarded or added.

Attention
CODEC

The configuration of codecs that can be used is made available in the call profile. Select the codecs that
will be used. You can drag them to define priorities, if there is more than one codec selected.

Cause Map
Defines the translation of causes of disconnection for different signaling types. When a disconnection is
received by one end of the call, the other end's cause map is used to translate that cause according to its signaling.
When the disconnection comes from a channel associated with the destination, the cause map will also determine
if a retry should occur because of the disconnection. Thus, the retry configuration is only valid for the source.
For example: For a call between an E1/T1 source and a SIP destination, if there is a disconnection from
the destination (SIP) (say a rejection of an INVITE), the cause provided by the SIP signaling will be used to determine
whether the source (E1/T1) should do a retry and, if not, which disconnection cause should be sent to disconnect
the source (E1/T1).

Field Description NAP available Version

for included
association

Cause of Number and name of the disconnection cause. All

disconnection causes have unique indexes.

Defines the value to be used to disconnect an ISDN

ISDN
signaling channel
Any -
Defines the value to be used to disconnect an R2
R2
signaling channel

Defines the value to be used to disconnect an SIP

SIP
signaling channel.

Enables or disables retry, when this cause is reported by

Retry Source 2.1.0
the associated channel as destination.

7. Configuring the Call Classification feature -

Analytics
You must have an Analytics license applied in the KMG to use this feature.
Note

Analytics is a call classification feature which, when applied in the KMG, allows for identification of
signaling during a call that is answered. It can indicate whether the call is answered by a human or not, allowing the
call center administrator to decide what can be done with this call.

Analytics performs automatic detection of voice on a network (VAD - Voice Activity Detection).

7. Configuration steps
The configuration of Analytics involves the following steps:

1. "Configuration of disconnect behaviors".

2. "Configuration of the call classification feature behaviors".
3. "Association of the call classification feature behavior to a call profile".
4. "Association of this call profile to a route that should be classified".

Configuration of these steps are described in the following sections.

 7. Configuring disconnect behaviors

"Configuration" → "Routing" → "Profiles" → "Disconnections"

Disconnect behaviors define which disconnecting code should be sent to the source channel when the
action performed by the call classification feature results in a "hang up".

1. To create a new disconnect behavior, click on the "Add" button.

2. Record the information as described in the following table:

Field Description

Name of the disconnect behavior. This name is displayed in the Analytics

Name
behavior creation step when the "Disconnect" action is selected.

Code that should be used for disconnecting when using SIP signaling. Values
SIP Code
between 300 and 699 are accepted.

Code that should be used for disconnecting when using ISDN signaling. Values in
ISDN Code
accordance with the ITU-T Q.850 standard are accepted.

3. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Configuring Analytics behaviors

"Configuration" → "Routing" → "Profiles" → "Analytics"

This Web interface displays the Analytics behaviors created and allows users to manage them.
Follow these steps to create a new Analytics behavior:

1. Click on the "Add" button. The Web interface will be displayed, as shown in the following image.
 You can also create an Analytics behavior by copying or editing a profile. Please refer to the "Configure
call properties" section, in this manual, for more information on call profiles.

2. Register a name and select the location where KMG will operate. The following table gives more details
on these fields:

Field Description Version

included

Name to be assigned to this behavior. This name will be displayed in the

Name -
"Call Classification" field of the profile register.

Select the country and language for signaling. The choice entered in this
field affects the set of defaults that will be displayed. For example: When
Location 3.2.0
choosing "Portuguese (Brazil)", the KMG displays the set of defaults for
this region and language
 3. In the "Timeout" panel, adjust the maximum time that the Call Classification feature will have to analyze
the audio of the call after connection is made. If the Call Classification feature cannot analyze the audio
within the time allotted, then the action set in Timeout will be applied.

Adjust the time for classification analysis carefully. Raising this value too high may increase
Attention the number of calls classified, but will raise the cost of telephony charges.

4. Define the action that the call classification should take in a timeout. These actions will also be
displayed for the signaling patterns in this Web interface. The following table describes what each
action does.

Field Description Version

included

The gateway will send a connection to the source and replace the
Answer
audio if it hasn't already been replaced.

Disconnection behaviors for selection. KMG will send an SIP or ISDN

code determined in the selected disconnection behavior. Please
Hang up
refer to the "Configure disconnect behaviors" section, described
earlier.

The gateway will play the selected audio on the destination channel
Play audio and hang
and, when the audio finishes or the destination end disconnects, it
up
will send a disconnect to the source similar to the "Hang up" action.

If the source is a VoIP channel, the gateway will send an SIP INFO
with the information obtained from the classification.
For example: -

X-Khomp-Analytics-CC: Fax
X-Khomp-Analytics-CC-Pattern: Fax 1100 Hz

Notify ● For the "Answer" action, the extension header will be sent
when the call is accepted.
● For the "Hang up" or "Play audio and hang up" actions, it
will be sent when the call is canceled or disconnected.
● For the "Notify" action, an SIP INFO will be sent with the
information.

Call Classification will ignore this classification. The analysis will

Ignore continue until another classification is found or until it reaches the
time defined in the "Timeout" panel.

The gateway will attempt to perform a call retry. If the retry is not
possible, the call will be terminated and a Log will be generated
informing the reason for termination.
Perform call retry 3.4.0
ATTENTION! - You need to enable the retry option in the
configuration of the source profile and allow it in the disconnection
cause map for cause 30003: Gateway analytics retry action, because
this setting takes precedence over analytics configuration.
 5. Configure the action (described in the previous table) for the audio and signaling standards that require
adjustment. The default action is that all calls are answered. Some signaling standards allow two types
of adjustments:

● Global Configuration: Analytics applies the behavior for any classified call with the default type of
signaling and its variations (if any).

● Specific Configuration: Analytics applies the behavior that is defined in each default standard variation.
If the signaling standard is disabled, the action defined in the global default standard is applied.

The following classifications are available:

Field Description

Voice mail Voice Mail detection; depends on the registered patterns.

Answering machine Automated answering followed by a recording and ending with a beep.

The signaling pattern indicates that the call has been intercepted by the
Interception signaling
carrier and that the service is not human.

The carrier is sending a message before the call connection is established;

this message is usually about the status of the destination number (out of
Carrier message
reach, off). Occasionally, a mobile phone voice mail can be identified as a
carrier message, as well.

Also detected by patterns. A classic example is Nextel's announcement:

Forwarding message
"Please hold, your message is being forwarded"

It is reported upon answering, when there was a voice before that, but no
Unknown message
pattern was recognized.

A beep informing that both the destination and origin numbers are from the
Portability tones
same carrier.
 Field Description

Used to differentiate the ringback tone (usually songs chosen by users), from
Calling tone
unknown announcements.

Answered by a person (short

Normal human response, for example: "Hello".
answer)

Answered by a person (long Business human response, for example: "Secretary of Environmental and
answer) Social Studies, Philomena, good morning".

Fax Fax equipment answering

Silence after answering. Its detection depends on the configuration of the

Silence
silence time-out interval.

6. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Transferring audio files to the KMG

If you select the "Play audio and disconnect" action, the "Audio file" field will be displayed so that you can
select an audio file.
These files are sent to KMF through the FTP protocol. Please refer to the "Access KMG via FTP" section,
in this manual, for more information on how to access KMG using this protocol.
The files should be uploaded to the /audio directory.
KMG accepts WAV files with the following specifications:
● PCM, 8-bit, 8000 Hz.
● PCM, 8-bit, 11025 Hz.
● PCM, 16-bit, 8000 Hz.
● A-LAW, 8-bit, 8000 Hz (RAW).
● U-LAW, 8-bit, 8000 Hz.
● GSM, VBR, 8000 Hz.

Audio files with unrecognized formats will be considered as being in RAW format (A-LAW, 8-bit, 8-KHz).

7. Associating the Analytics behavior to a profile

"Configuration" → "Routing" → "Profiles" → "General"

After creating Analytics behavior, you must associate it to a call profile. You can edit an existing profile
to attribute an Analytics behavior to it, or create a new profile by copying an existing one.

1. Click on the "Add" button to create a new call profile, or on the " " icon to copy the properties of an
existing profile, or click on the " " icon to edit a profile that you have created previously.
2. Select an Analytics behavior that was created previously, as described in the "Configure Analytics
behaviors" section.
 3. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

Please refer to the "Configure call properties" section, in this manual, for more information on call
profiles.

7. Associating a profile to routing

This is the final step in the configuration of Analytics. For KMG to run the Call Classification feature you
must associate the call profile containing Analytics to the routing where the classification is needed.

1. Access the menu "Configuration" → "Routing" → "Routes".

2. Create or edit a route where call classification will be applied.
3. In the "Destination profile" field, select the profile with the Call Classification feature applied, as
described in the previous section, "Associate Analytics behavior to a profile".
4. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Associating the classification feature to the SIP

trunk, E1/T1 trunk and GSM channels
You can also associate an Analytics call profile to an SIP trunk, E1 trunk and group of GSM channels

1. Go to the "Configuration" → "Routing" → "NAPs" menu.

2. In the table displayed, click on the "New NAP" button to create a new NAP, or click on the icon "
" to edit an existing NAP.

3. In the field "Profile", select a call profile that contains an associated Analytics behavior.
4. Click on the "Save" button.
5. Access the menu "Configuration" → "Routing" → "Routes".
6. 6. Create or edit a route where call classification will be applied.
7. In the field "Destination Profile", select the option "Use NAP profile".

7. Setting up an operating period for routing

"Configuration" → "Profiles" → "Operating period"

Displays the time restrictions that can be applied to routes. This feature allows you to define that a
particular route operates only during business hours, for example. Or that a route operates only on weekends. Thus,
it is possible to configure overflow to another valid route if the main route is outside the defined hours.
The KMG has some default restrictions, which can be removed or edited by clicking on the action
buttons in the "Options" column.
To create a new restriction, click on the "New Time Restriction" button. You can define a restriction by
time, day of the week, day of the month, months, or by a combination of these restrictions. It is possible, for
example, to create a restriction on which a route will operate on Mondays and Wednesdays from 8 am to 6 pm.
Finally, click on the "Save" button and then on the "Apply" button for the changes to take effect.

In order for the time constraint to work correctly, KMG must have the date and time set
Attention correctly.
 7. CDR - Call Detail Record

CDR is a register that contains the details of all calls processed by KMG. The following list contains
some examples of information included in the CDR:

● The phone number of the subscriber originating the call (calling party, A-party).
● The phone number receiving the call (called party, B-party).
● Additional digits on the called number.
● The starting time of the call (date and time).
● The duration of the call.
● The identification of KMG device.
● The results of the call, indicating, for example, whether or not the call was connected.
● The type of technology used in the call (VoIP, SMS, etc.)
● Any fault condition during call.

The registers can be visualized in KMG Web interface, at the "Diagnostics" → "CDR" menu, and can be
sent to a RADIUS server or to an FTP server.

7. Customizing and activating a CDR

1. Access the menu "Configuration" → "CDR" → "Text file".
2. Enable the "Text file" option.
3. In the "Rotation Period" field, select one of the following options:

● per minute: One file will be generated per minute containing all calls finalized during this minute.
● per hour: One file will be generated per hour containing all calls finalized during this hour.
● per day: One file will be generated per day containing all calls finalized during this day.

Call records are only written to a file when the call is ended. This way, the complete call data will always
be in a single file. If CDR file generation has been defined as per minute, for example, and a connection is in
progress at the end of the CDR file generation minute, the record of this call will be written to the next file generated,
if it is finalized during the next minute.
 4. In the "Recording Style" field, select the style in which the CDR file will be recorded on the server:

● One line per routing attempt: One line will be written to the file for each routing attempt, including "retry".
● One line per call: One line will be written to the file for each call made.

5. In the "Formatting" text box, customize the CDR record. You can enter any desired text; all special fields
(${field_name}) will be replaced with the logged call data. The available fields are listed to the right of the
text box and can be added by dragging them into the text box or by double-clicking on the desired field.
Before dragging the field into the text box, you must position the mouse cursor where you want to insert
the field.
It is advisable to use an operator between the fields to make the file easier to read, such as a comma (,),
semicolon (;), or whatever provides better visibility. Note that the date and time fields have several
formatting options. There are some examples available in the interface help feature, located in the lower
left corner.
The available fields have a help button, represented by a "?", which contains their description. To view it,
position the mouse cursor over the help icon.

6. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Sending a CDR to a RADIUS server

"Configuration" → "System" → "CDR" → "RADIUS"

RADIUS servers can be configured to collect information from the calls processed by the gateway for
registration and billing analysis. This configuration Web interface allows you to send KMG CDR data to up to 3
RADIUS servers.
To activate this feature, enable the RADIUS option and add a server. You can add up to 3 RADIUS
servers.
The description of each field is verified as follows:

List of Servers

Field Description

Address IPv4 are accepted as valid addresses

Port RADIUS server port. To set the default port (1813), use the value 0.

Value to be shared between the server and the client (get more information
Shared secret
at RFC 2865).
Advanced options

Field Description

Maximum time the KMG should wait for a valid response from the RADIUS
Maximum Request Wait Time server. This parameter works in conjunction with the "Maximum number of
retransmissions".

Maximum number of times that the KMG should retransmit a packet to the
RADIUS server. This parameter works in conjunction with the "Maximum
Request Wait Time". If there is no response from the server after the
Maximum Number of
maximum wait Time of the last retransmission, then the next server must be
Retransmissions
used. If there are no more servers configured or all are already inactive, the
packet will be discarded. The server may be marked as inactive if the
deactivation time is configured.

Defines how long a server should be marked as disabled due to transmission

Server Deactivation Time Due to failures. During this period, no further attempts will be made. If all the servers
Failure are disabled, the packets will be discarded. Use the value 0 (zero) to never
disable the servers.

Date Format Date and time format expected by the RADIUS server.

Parallel Requests Number of parallel requests that should be made to the RADIUS server.

When the submit queue is longer than the value specified in this field, new
Acct-Status-Type packets with a value of Start (1) will be discarded, to allow
Submit Queue Size for the more Acct-Status-Type packets with Stop (2) to be added to the queue, until
disposal of packets the submitting queue reaches the "Maximum Queue Size" value. The
Acct-Status-Type: Start (1) "Maximum Queue Size" setting takes precedence over this setting. Therefore,
you can either use the same value for both or a smaller value in this field, if
you want the queue to have only the maximum limit.

When the submit queue is longer than the value specified in this field, new
Maximum Queue Size
packets are discarded.

7. Sending a CDR to an FTP server

"Configuration" → "System" → "CDR" → "Maintenance"

This interface presents the options that allow you to send CDR records to the FTP server of your choice.
 1. Enable the "Automatically send to FTP server" option.

2. Register the information from your FTP server in the "Server Address", "Server Port", "User", "Password",
and "Destination Directory" fields.
3. Click on the Save button. KMG will test the connection with the FTP server. This operation can take a
few minutes.
4. Finally, click on the Apply button for the changes to take effect.

After you click on the "Save" button, KMG will perform a connection test with the FTP
Note server. During the test, no response is displayed on the Interface. After the test is
complete, the test result will be displayed.

7. Setting up the CDR retention time in KMG

"Configuration" → "System" → "CDR" → "Maintenance"

By default, CDR records are stored in KMG for a period of 30 days. After this period, the older records
are removed. This configuration is not related to
In this Web interface you can change the retention period. CDR storage is also limited by the capacity of
the KMG hard drive. The KMG 200, KMG 400 and KMG 1600 offer 2.5 GB. The KMG 3200 offers 8 GB.

7. Configuring portability

"Configuration" → "Telephony" → "Portability"

The portability feature, when enabled, is used in call routing to identify the carrier of the destination
number, so as to use the lower cost route.
When you enable this option in the system, the "Check portability" option will be available in the route
configuration ("Configuration" → "Routing" → "Routes"). When you select it, two more options will become available:
"Valid service providers" and "Change number for portability query".
When the portability query is enabled, it will check each call to discover which carrier the destination
number belongs to and the result will be stored. If the carrier of the destination number is listed in the "Valid service
providers" field, this route will be used, otherwise the routing will test the next routes.
In the "Valid service providers" field, you must select which carriers can be used with this route.
The "Change number for portability query" field is used to format the number to be checked, according
to the format expected by the query service provider. For example, if the dialed number is 0994899998888 and the
provider expects the Area code + number format, you will need to use a regular replacement expression, such as:
/0..([0-9]*)$/$1/, so the first 3 digits will be removed and the query will be performed using 4899998888.

7. Portability via Webservice

To enable the portability query via Webservice, you must enter the data of the portability query service
provider and import the set of carriers that can be returned by the portability service (RN1). This set must be
provided by the service provider.

Go to the topic "Importing data" for details on how to import RN1 codes.
 Field Description

URL of the portability service. Use the wild card ${number} to indicate where
URL
the number being checked should be entered.

Method Configures the HTTP method used for the request.

You must configure a POSIX-type regular expression, which will be used to

extract the carrier code from the response provided by the service provider.
Response pattern
Keeping this field blank means the response provided contains only the
carrier code.

Time-out Determines the maximum wait time for the response.

Parameters submitted using the POST method. Use the wild card ${number}
Additional Parameters
to indicate the key that should contain the number being checked.

If configured, it should contain the HTTP-BASIC type authentication user. It

User
also depends on the Password field being configured.

If configured, it should contain the HTTP-BASIC type authentication

Password
password. It also depends on the User name field being configured.

Error handling Enables or disables special handling of errors.

It determines the number of errors that the portability system should accept
Acceptable amount of errors without taking action. When the defined amount is reached, the portability
system must be disabled according to the defined Time-out in case of errors.

Defines the time (in seconds) during which the portability system will be
Time-out in case of errors temporarily disabled if the acceptable number of errors is reached. After this
time, the service is enabled again.

7. Portability via Local Database

Enables the query using an internal KMG database. In order for this service to work properly, you must
import the set of valid carriers, the set of numbers and/or the set of prefixes.
When this service is enabled, the query will be performed in the following order:

1. Checks if the number is defined in the set of Ported numbers.

2. Checks if the first "n" digits of the number are defined in the set of Prefixes, where n is the size of the
prefix.

7. Importing data
To import the data, open the import wizard using the "Import data" button and then fill out the form. The
import form applies to both Webservice and local database portability.
The submitted file must be a text file or compressed text in "gzip" format. If the file has special
characters (accents, for example), you must use UTF-8 encoding. The file must be in the format of fields separated
by a well-defined character, with one record per line (mandatory). The CSV format is recommended for the text file
provided. Empty lines are ignored.

During import, the portability service will be disabled.

Note
The import form has the following fields:

Field Description

File File to be imported

Select the type of data in the file. For portability via Webservice, only the
Type of data in the file
Carrier data type is available.

Select the data type in each field according to the order in which they are
Format of the fields in the file
displayed in the file being imported.

Field delimiter character Character that must be a recognized field separator.

A character that must be recognized as a field encapsulator. When this

Field encapsulation character character is found, there must be another equal character for value
recognition.

If enabled, it deletes all records that match the "Type of data in the file"
Remove all records before
selected before initiating the import. "Caution", because dependent records
importing
will also be erased (Numbers and Prefixes are dependent on the carriers).

If enabled, it skips the first line of the file. This option exists for compatibility
Skip first line
with having columns headers in the first row.

If enabled, the "Operation" field, informed in "Format of fields in the file", will
Incremental Import
be used as a guideline to what to do with the record.

If enabled, invalid rows (which could not be imported) will be discarded and
Skip invalid rows logged. If disabled, the import will end with an error on the first invalid line
encountered.

We recommend carrying out the first import of the portability database using the following order:

1. Carriers.
2. Ported numbers (local database only).
3. Prefixes (local database only).

7. Scripts

"Configuration" → "Routing" → "Scripts"

Products in the KMG line have support for Scripts in the LUA format. Lua is a programming language
designed to extend applications. It allows procedural programming, object-oriented programming, functional
programming, data-oriented programming, and data description. More information on LUA can be found at the site
www.lua.org.
The use of a script basically allows you to make changes to the SIP header, and change the source and
destination numbers. You can manipulate the same information that is logged in the CDR file.
The script is associated with a NAP. When the route has a scripted NAP, then it executes its instructions.
The script can be run on both a source NAP and destination NAP.
 7. Adding a script
1. Click on the "New Script" button.
2. Enter the script name.
3. Write the script in the displayed text area.
4. Click on the "Save" button and then on "Apply."

The text area has instructions for manipulating call data in the KMG. The script must contain the
following functions so that it runs correctly.

function prerouting()

return true
end

function postrouting()

return true
Assigning a script to a NAP
end
To associate a script with a NAP:

1. Access the "Configuration" → "Routing" → "NAPs" menu.

2. Create or edit a NAP.
3. In the "Script for Routing" field, select the desired script, which was previously added.
4. Click on the "Save" button and then on "Apply."

7. Managing SIM cards

"Configuration" → "Routing" → "SIM Cards"

This option is only displayed if KMG has a GSM license.

Note

This Web interface displays the SIM cards inserted in the GSM interfaces along with their properties,
either through internal modules (KMG 400) or external telephony modules - KMG Module. When you insert a
properly registered new SIM card, it will be displayed on this Interface, so that it is associated with an existing GSM
NAP.

The icon " " indicates that a new SIM card was discovered.

By clicking on the icon " ", you can associate a new SIM card to a group of GSM channels that have
been previously registered and also include or edit the number of the SIM card. This number will be used as the
destination number in routing.
 7. Associating a new SIM card with a group of existing
GSM channels - NAP GSM
When inserting a new SIM card, you can associate a group of previously created GSM channels. See the
section Configure GSM channels for more information on the creation of a group of GSM channels.

To associate a new SIM card to a NAP GSM:

1. Click on the icon " " for the SIM card that will be associated with a GSM group.
2. In the column "NAP", select the group of GSM channels created (NAP GSM). Repeat this action if there
are more SIM cards that will be associated with a NAP GSM.
3. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Adding a SIM card manually

You can add SIM cards before installing the SIM cards in the GSM interfaces connected to the KMG.
This way, when a SIM card is inserted, it can be associated with a NAP GSM defined automatically.

To add a SIM card manually:

1. Click on the "Add Manually" button.

2. Register the SIM card ICCID. This information should be made available by your GSM telephony carrier.
3. Register the SIM card number. This action is optional.
4. In the column "NAP", select the "NAP GSM" previously registered.
5. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. Creating SMS routes

The SMS routes are the associations between the KMG and a destination GSM NAP, which will be used
to allocate the GSM channel for sending. You can also define filters for its use, such as the destination number and
a portability query.
For each SMS sent, the routes will be evaluated in the order of priority defined in the configuration. The
first route that satisfies all filters will be used.
To send and receive SMS using KMG, an SMS API is available.

Field Description
SMS messages can be sent even if the channel is occupied with a voice call.
Name Note Name of the SMS route. The name must be unique among all routes.

The priority with which this route will be evaluated. Lower priority routes will
Priority be assessed before higher priority routes. There is no defined order for equal
priority routes.

Destination Number Filter by POSIX.1-2001 regular expression for the SMS destination number.

Check Portability If enabled, it will indicate the use of the portability query for this route.

When the option Check Portability is enabled, it selects the service providers
Valid service providers
that are valid for this route.

Change number for portability Number to be used for the portability query. Accepts regular expressions
query (POSIX.1-2001).

The NAP to be used for the allocation of the outgoing SMS channel, only
Destination NAP
GSM NAPs are displayed.
 7. SMS sending test

With the KMG, it is possible to carry out SMS sending tests to ensure that the configured SMS routing is
working properly. For this, you must use the sending form, providing information on the required fields "Destination
Number" and "Text Message". You can also request the sending confirmation.
When sending, the following information is displayed: the value of the generated sms_reference code,
the ICCID of the Sim Card that was used to send the SMS, the route used, and the carrier resulting from the
portability query, if any of the evaluated routes use portability query. Confirmation of successful sending (by the
carrier), as well as of receipt, if requested, can only be done through the Logs or the notification system.

7. Configuring the reception of SMS notifications

"Configuration" → "Routing" → "Notifications"

The "Notifications" option is displayed only if KMG has an SMS telephony interface.
Note

With a GSM telephony interface, the KMG can send the content of SMS messages received to HTTP
servers that are able to receive GET and POST type requisitions or even REST APIs.
This feature makes it possible to integrate the KMG with systems that process SMS messages received,
such as a system that solicits a response from a client by SMS, for example.

The following types of messages are sent:

● SMS sending errors: Errors issued by the telephony carrier when a message is not delivered.
● SMS message received: Messages sent by a person, for example.
● Confirmation that the SMS message was sent: Confirmation of the sending of a message. This option
should be activated in the SIM card.

To use this resource, an HTTP server must be registered first. It is possible to register more than one
server to receive different types of messages separately, or send all types to the same server.

1. In the "Transport" panel, click on the "Add" button.

2. A form will be displayed to register the server that will receive the messages. A description of the fields
follows:

Field Description

Used to describe the server. The panel "Default transport for types of notification
Name
will display for selection

User of authentication of HTTP service, or API REST. This field is not mandatory in
User for authentication
the event that the server does not request authentication
 Password for authentication of HTTP service, or API REST. This field is not
Password for authentication
mandatory in the event that the server does not request authentication

Remote host or IP HTTP server address

Remote port HTTP server port enabled to receive this protocol

Remote path HTTP server directory where the messages will be stored

Method used to send messages: GET or POST. If the method chosen is

Method POST, the columns "Name" and "Value" will be displayed to register this
information

3. Click on the "OK" button.

4. If it is necessary to register another server, repeat the previous steps.
5. With a server registered, you can choose which server will receive each type of message. In the panel
"Default transport for types of notification", select the server registered in the column "Transport".
6. Click on the "Save" button. If no other configuration is made, click on the "Apply" button for the changes
to take effect.

7. High Availability - HA

Resource available on all KMG models, however HA is not supported by the EBS
Note internal model, if present.

If the main hardware fails, another KMG of the same model will take on the gateway functions,
preserving network settings, including the IP addresses. The High Availability feature was designed to operate with
or without external telephony modules.
A KMG installed with High Availability offers "system security" and guarantees "uninterrupted continuity
of operation".

7. Network connection for HA

A KMG with HA can use any network port for this feature by using it to connect the two pieces of
equipment, provided that it is configured for this.
Set an IP address for the maintenance interface that differs from KMG IP address for regular interfaces.
This IP will be used for administration and individual access to the equipment, however these should be separated
by Master and Spare. The IPs of the remaining ports will be shared among the equipment, with only the active
equipment responding for them.
By sharing the same IP address, you can connect the KMG to a redundant network structure, and thus
ensure its operation even when one of the network structures is down.
 7. High Availability with external telephony modules

Khomp's external telephony modules have two network interfaces. When using this module in an HA
scenario, both the master KMG and the spare KMG must be connected to the same module and the same gateway
network interface. For example: If the master KMG is connected to an E1 telephony module via the P0 interface, the
spare KMG must also be connected to this same module via the P0 interface, as shown in the previous figure. The
same rule should apply to other network interfaces and modules, if available.

7. Configuring High Availability

Enabling/disabling the High Availability feature is a task that must only be performed by
Attention Khomp’s support team.

"Configuration" → "Network" → "Interfaces"

With high availability licenses loaded on the equipment, the configuration options become available.
Initially it is necessary to configure the Master equipment selecting the network interface that will be used for
management, as well as its IP and network mask. The same process is repeated afterward on the Spare equipment.
At that point, both pieces of equipment will be operating in high availability.

Field Description

Determines the network interface for gateway management to which these

Network Interface
settings will be applied.

The IP address that will be used to access the KMG. After enabling High
Administration IP
Availability, this will be the actual IP address of the gateway.

Subnet mask that will be used to access the KMG. After enabling High Availability,
Subnet mask
this will be the actual subnet mask of the gateway.

7. High Availability operation

When this feature is enabled, the IP address set in the High Availability option becomes the actual IP
address of the KMG. Access to the Web interface will occur through it. All the network settings applied in the
Network gateway menu becomes virtual and is replicated to the spare gateway, as well as the other settings. Any
changes to the KMG settings will be replicated to the spare gateway.

7. Verifying the master and the spare gateway

The spare KMG will not have access to the configuration menus, although the configuration will be visible,
and only the options for high availability can be altered.
The "Monitoring" → "System" menu also displays the master (main) and spare (redundant) KMG modules
via the serial number of each gateway. In addition, information about heart-beat will be displayed.

8 Monitoring
"Monitoring"

The Monitoring interface should be used to monitor the KMG operation. In this section you can check
the created NAPs and track the status, analyze the KMG telephony devices and modules and their respective
channels.
In the "Refresh rate" field, you can define a period for the interface to update the information
automatically, according to the defined time. Or update manually.
 8. NAP
"Monitoring"→ "NAP"

List of information for monitoring of one or more NAPs. The columns provide the following data.

Field Description

NAP Provides the NAP name -

Shows the NAP type, which can be SIP, Trunk

Type -
or GSM. Depending on the KMG model.

● Active
Status Provides the NAP status for its use.
● Inactive

● Free: The channel is free to make

or receive calls.
Provides the status of the channels ● Failure: The channel is faulted,
Channels
associated with the NAP. unable to receive or make calls.
● Busy: The channel has an
incoming or outgoing call

● SIP: It will display registration

information and Keep Alive
options configured for the SIP
account, if applicable.
● Trunk: No information is provided
Provides details about the NAP. For each NAP
Details ● GSM Information is shown and it
type, distinct information can be provided.
allows for editing and controlling
monthly use of minutes and SMS
text messaging, if the NAP has
this configuration enabled in the
shared format.

8. SIM cards
"Monitoring" → "SIM Cards"

This feature is only available on devices with a GSM interface.

Note

The status information of the SIM Cards configured in the system is displayed. You can also change the
minutes and SMS message control for SIM cards that have this setting enabled.

The columns provide the following information:

 Field Description

Displays the ICCID information of the SIM

SIM Card card, in addition to the IMSI and carrier -
information, if it is registered

Informs the NAP to which the SIM Card is

NAP -
associated

● Active: SIM card available for

use
● Inactive: Unavailable due to
Status Provides the SIM Card status for use
lack of registration,
configuration, usage control or
other factors.

● Free: SIM card available to

receive or send SMS
messages.
● Sending SMS message: SIM
card sending SMS
● Receiving SMS message: SIM
Provides the status of the channels for SMS
SMS Status card receiving SMS
messages.
● Limit Reached: Limit of SMS
usage has been reached; in this
status the SIM Card is
unavailable to send SMS
messages; it only allows
receiving SMS messages.

Displays the remaining minutes for call

Minutes generation if usage control is enabled. You
-
remaining can edit this value if shared usage control is
disabled.

Minutes Informs the monthly Minutes configured for

-
configured this SIM card

Displays the amount of SMS messages

SMS messages remaining if usage control is enabled. You can
-
remaining edit this value if shared usage control is
disabled.

Configured SMS Informs the number of monthly SMS

-
messages messages configured for this SIM card

Displays details about the SIM card, informing

Details its Device and index, as well as its last status -
update.

8. Devices
"Monitoring"→ "Devices"

It displays monitoring information on the physical devices - Boards and external telephony modules
(EBS) - and Logical devices (VoIP). For each device the following is displayed:
 Field Description Values

It is the device identification within the

system, represented by a number from
ID -
0 to (N), N being the number of devices
configured in the system.

This is another way to identify a

device, through the serial number
Serial assigned to a single device. Khomp -
does not distribute two devices with
the same serial number.

Model It is the product name. -

● Up – The device is connected,

configured and functional.
● Down – The device is configured, but
not connected; therefore, it is not
functional.
Indicates whether a device is active or ● N/A – Information not available. This
Status
inactive. situation occurs when it was
impossible to retrieve the information,
probably because one or
more Services that are essential for
the system operation are not active.

Overview of the activities of the device

channels in a given moment. It
Channels includes the number of free, in fault, or -
on call channels at the moment the
screen was refreshed.

Displays the status of devices with an

Links E1/T1 link. Other telephony interfaces -
will not display data in this column.

Displays a chart showing current call

traffic statistics for the telephony
● Blue - completed calls
module. This information changes as
● Red - failed calls
the screen refreshes. On the right
● Orange - missed calls
Options column you can see the meaning of
● Gray - no data to display. There was no
the colors shown in this chart. You can
call traffic.
click on this chart for more details.
Refer to the "call chart" item for more
information
 8. Status
When the operation is normal, all devices must be in "Up" status, as well as all their E1/T1 links.
Some possible causes for a device to appear as "Down" or "N/A" are:

● It is not connected to the network.

● It is not properly powered.
● It is defective.
● It is not properly configured.
● It is configured in another server.

8. Links
"Monitoring" → "Links"

This feature is only available on devices with an E1 telephony interface.

Note

The purpose of this section is specifically for monitoring digital links. All the links configured in the
system are listed, as well as any information about their current situation. The following information items are
available:

● Link: The links are identified in the "Link" column by a code consisting of the serial number of the device
and the link number in the device, and the assigned name (optional), separated by the character ".".
● Status: Operation situation of the links. It indicates if a link is active or not, according to the following
values:
○ Up – the link is aligned, configured, and functional.
○ Down – the link is not operational. The "Alarms" column has information about the cause.
○ N/A – Information not available. This situation occurs when it was impossible to retrieve the
information, either because one or more Services that are essential for the system operation
are not active, or because the device to which this link belongs is not connected.
● Alarms: Information about link errors. When the link is operational (Status="Up"), only the character "-" is
displayed.
The possible alarms are:

Alarm Meaning Possible causes

Signal loss Rx circuit interrupted Unplugged cable

Cable connected, remote equipment on and not

Network alarm Standard signal reception all-1
configured

FrameSyncLost Loss of frame alignment Cable connected, but remote equipment off

Remote equipment configured with incorrect type

MultiframeSyncLost Loss of multiframe alignment
of signaling

● Tx circuit interrupted
Alarm was issued in remote ● CRC4 not configured
Remote alarm
equipment ● Internal failure in the remote
equipment

HighErrorRate Reserved -

Unknown alarm Reserved -

E1 Error Error in the E1 controller Link framer (hardware component) is defective

NotInitialized E1/T1 not initialized Device not configured

Device Unreachable Device is "Down" Check item: "Status"

● Variations: It indicates whether there are variations in the link error counters. For example, if a cable
presents a bad contact issue at a given moment but works well after that, the counters will indicate that
a problem occurred.

If there are no variations (all counters are reset), clicking on the "View" link will display a table of
variations without error increments. Otherwise, if there is a counter with a value greater than zero, the link will
appear in bold.
Additionally, there is the option to reset the counters through the "Clear" link.
The following counters are monitored:

Alarm Meaning Possible causes

Presence of a critical alarm (SignalLost,

Blocked Link failure count NetworkAlarm, FrameSyncAlarm or
MultiframeSyncAlarm) Check the Alarm item.

Signal loss Signal loss alarm transitions Check Alarm SignalLost

Alarm notification Network alarm transitions (all-1) Check Alarm NetworkAlarm

Frame alarm Frame alarm transitions Check Alarm FrameSyncLost

Multiframe alarm Multiframe alarm transitions Check Alarm MultiframeSyncLost

Remote alarm Remote alarm transitions Check Alarm RemoteAlarm

Slip alarm Number of discarded frames Links operating with different clocks

PRBS Reserved -

Incorrect E-bits Number of incorrect E-Bits received Incorrect configuration of CRC4

Jitter variation Reserved -

Frame time without

Reserved -
synchronism

Frames without
Reserved -
synchronism

● Presence of frame alarms

Number of incorrect frames (zero
Frame errors ● Links operating with different clocks
time slot) received
● Insufficient grounding for the device

● Grounding jumpers not connected

Number of bipolar violations (HDB3) ● Insufficient grounding for the device
Bipolar violation
received ● Physical circuit has problems
 ● Incorrect CRC4 configuration
Number of incorrect CRC4 frames
CRC-4 error ● Insufficient grounding for the device
received
● The physical circuit has problems.

● The device is not connected

N/A Information not available ● service required for the system
operation is not active

● Signaling: It displays the telephony signaling protocol used on the link defined in E1/T1 Links
"configuration".

8. Restarting
It power cycles the link framer, causing a physical restart of the system. All the calls are dropped and
the link is configured again.

8. Blocking
It power cycles the link framer, causing a physical restart of the system. All the calls are dropped and
the link is configured again.

8. Channels
"Monitoring" → "Channels"

This section gathers monitoring information about the device channels. To view the channel list, select
the device and then the channel range. In devices with digital links, channels are grouped in links.

The channel list provides the following information:

● Channel: Channel index within the device, a number from 0 to N-1, where N is the number of channels
available in the device.
● Signaling: Telephony signaling protocols associated with the channels.
● Status: It informs the state of a certain channel. If the channel has a problem, the causes are described
in the "Details" column. The possible values are:

Value Meaning

Idle The channel is free to make or receive calls.

Fail The channel is faulted, unable to receive or make calls.

On Call The channel has an incoming or outgoing call

● Details: It provides more detailed information on the channel status for each type of interface.
According to the following table:

Interface AddInfo Interface AddInfo

● Free ● Idle
● Incoming ● Call in Progress
● Outgoing ● SMS in Progress
E1
● Fail GSM ● Modem Error
● Locked for Outgoing ● SIM Card Error
● Locked for Incoming ● Network Error
● Remote Lock ● Not Ready

Interface AddInfo

● Free
VoIP ● Outgoing Lock
● Incoming Lock
 ● Dialed Number: This field informs the called number in the current call.
● Duration: Duration of current call.
● Average time: This field informs the average time duration of the calls made.
● Statistics: It informs the statistics of calls computed by API K3L.
● Incoming: Total of calls received.
● Outgoing:
○ Amount: Total number of outgoing calls.
○ Failures: Unsuccessful outgoing calls.
○ Completed: Successful outgoing calls.

8. Monitoring GSM channels

"Monitoring" → "Channels"

This feature is only available if KMG has a GSM telephony interface.

Note

● SIM Card: In this field, you can select the channel index that will be used by the GSM modem, with the
option of alternating between the two existing slots, 0 and 1.

A request for changing the SIM Card has precedence over the automatic rotation by use and
can be configured in the NAP.
Attention If there is a call in progress using this SIM Card, it will be finalized.

● Carrier: It informs the carrier name, registered in the ERB cell, in which the modem is currently registered
at the Level of RX signal, as a percentage, received by GSM antenna. The signal strength can range from
-113dbm (1%) to -51dbm (100%).

8. Statistical Call Chart

The monitoring of devices, links (E1/T1) and channels have charts that show the statistics of call traffic
in the telephony module. To access, just click on the chart located on the bar in the "Options" column.
The information displayed in the chart is from the moment it was opened. To update the data, just click
on the "Update charts" button.

Four pie charts will be displayed:

8. Device statistics
Displays the device information since the last KMG boot.

● Dialing completion: Percentage of completed, unanswered and failed calls, i.e., calls that for some
reason could not be completed.
● Cause of failures: If there are failed calls in the previous chart, the percentage of these causes will be
displayed.

8. Status of the channel

Displays the device channel information at the moment the chart was opened, i.e., channel data at the present time.

● Occupation: Percentage of free, busy, or faulty channels.

● Incoming and Outgoing: If there are busy channels in the "Occupation" chart, the percentage of incoming
calls and outgoing calls will be displayed.

8. Links
If the telephony device has an E1/T1 link, you can view the dialing completion of all links available on the
device, which may help identify possible failures in these links.
 9 Diagnostics
The diagnostics area allows for reading and downloading system logs, as well as changing log levels.
The most important or urgent system messages are shown on the initial Web interface (Summary). The content
shown is the same as the messages.log file.

9. Downloading logs
In addition, in this screen you can perform a download of the latest logs of the system. Downloading the
latest Logs makes it easier to provide a packet with all system logs to the technical or diagnostics team using tools
of the workstation currently being used to access the Web interface.

9. Understanding the log messages

All messages have some default elements that help in their identification and origin. The most
important among these elements are:

● The first character indicates the priority level of the message and can be (in order of higher priority to
lower priority):
○ E - Error - Error messages usually indicate inability of one of the services to properly operate.
○ W - Warning - Serious problem warning messages usually indicate a behavior that will not
disable the service operation, but could compromise the expected result.
○ I - Information - Informative messages usually indicate start and end of service execution,
as well as start and end of device operation. This level of log is displayed on the log
summary screen.
○ N - Notify - Notification messages contain relevant information, however, they are normally
not prejudicial to the operation of the system.
○ T - Trace - Debugging messages, which should always be disabled during normal operation,
provide service debugging data, normally useful when diagnosing an operation problem.
● The date and time in which the message appeared will always be between the first and the second
vertical lines "|";
● From the second vertical line "|" to the parentheses at the end of the line, the formatting varies according
to the service;
● The name of the service that issued the message will be located at the end of the line, enclosed in
parentheses.

Messages labeled as error, warning and information priorities are recorded in the service Log and in the
messages.log file.

The following are some common conventions used by the services:

● Identify the device by using the letter D followed by the device serial number.
● Notify the channel using letter C followed by the channel number.
● Notify the link using letter L followed by the link number.

9. Changing register levels and diagnostics mode

"Diagnostics" → "Options"

In this area you can enable the diagnostics mode, and change the Log levels.

9. Diagnostics mode
It is not recommended that the diagnostics remains enabled during normal device
operation. Whenever debugging messages are unnecessary, this mode should be
Attention deactivated.
 9. Advanced options
Allows you to enable or disable the trace (debugging) of the Logs displayed in this option. Other
information, such as information, warnings and errors, will continue to be recorded in the Logs.
The first record, "Logs", has the "FullLog" component that overlaps the trace registration in the other
logs, so it is not necessary to enable each log individually.
When you expand the log types, the "Value" is displayed in the first line. This setting affects all other
components of the Log. Therefore, when you set the K3L Log value as True, the system understands that all other
components of this Log will also be true, except if a particular Log is marked as Never. The following explains some
more details on log debugging levels.

● False: The Log (trace) debugging is not recorded in the Log. However, if the Value or FullLog fields are
set to true, this value is overridden.
● True: The Log (trace) debugging is recorded in the Log, even if the Value or FullLog fields are false, this
field remains true.
● Never: The debugging of this Log will never be recorded, even if the Value or FullLog fields are set to
true.

9. Logs
"Diagnostics" → "Logs"

It provides access to the service Log files, in which the following Logs stand out:

● messages.log - The most important and urgent Logs are also saved in this file.
● k3l_intf.log - Debugging Log of the received commands and sent events from/by the API to the other
services.
● kgateway.log - Routing service Log.
● kmp.log - VoIP service Log.
● isdn.log - ISDN debugging Log. Contains the option for analysis that displays the log in the most
user-friendly format.
● r2.log - R2 debugging Log. Contains the option for analysis that displays the log in the most
user-friendly format.
● voip_msg.log - SIP debugging Log.
● 9. Config
gsm.log - GSM debugging Logs.

This option displays the KMG configuration files with the date of the last reboot of the machine.

9. Licenses
This option displays the licenses applied to the KMG. You can also view and download the license file as a backup.

9. OLD
Old logs are automatically moved to the subdirectory /old, where they are archived with the date and
time on which they were moved, subsequently compressed and eventually removed, depending on disk usage.
The system is configured to retain up to 5GB (1 GB on the KMG 200 model) of logs, combining all active,
old and compressed logs. When this number is reached the logs are automatically removed, from the oldest to the
newest.

9. CDR
"Diagnostics" → "CDR"

The CDR file is a list of records of calls made by the KMG. For each call made, a record ticket is
generated to store the call data.
In the CDR diagnostics environment it is possible to download the CDR file or view it directly on the Web
interface.
 9. Packet capture
"Diagnostics" → "Packet capture"

It allows you to enable network packet capture on the gateway. This capture file uses the pcap
compressed format, employing the gzip algorithm. We recommend using the free software Wireshark for viewing
the capture files.
The capture files are stored in the /capture directory in the FTP and are also accessible through the Web
interface, at this page.
Once the capture is started, it can be stopped by any of the following three conditions:

● It was requested that the capture be aborted, through the Web interface.
● The configured time limit was reached.
● The maximum size allowed has been reached.¹

¹The maximum size is determined by KMG model and is shown in the help section for the "Duration" field. The
following configurations are shown:

Field Description

Duration Time limit for capture duration (in minutes).

EBS modules If enabled, performs packet capture of the telephony module.

If not defined, it captures packets from or to every host.

If defined, it captures only packets from and to the defined hosts.
Filter by host
Addresses from the gateway itself can be used to filter traffic by network interface,
on models with multiple interfaces.

9. Audit
"Diagnostics" → "Auditing"

Routing audit refers to audio recording during the call classification period, i.e., the period between the
beginning of the call and the result of the classification.
When the result of the classification occurs, the audio is stored in a folder accessible through the FTP
protocol with the name of the classification result, so this audio can be used to verify that the classification is
correct. To enable auditing, the destination NAP of a given route must be associated with a configured Analytics
profile.

Only the channel allocated to the destination NAP is recorded, as this is the channel
Note being evaluated by the classification feature.

To enable auditing, the destination NAP profile must have Analytics behavior selected. Routes that meet
these requirements will display a tick box to the left of the route name. Additionally, it is necessary to configure the
displayed fields:

● Duration: Configures how long, in minutes, the audit process should record calls.
● Record the entire call: When selecting this option, it will record until the call is ended, otherwise it will
record until the end of the classification.
● Routes to be audited: Defines which routes should be audited. You must enable at least one.

You can only enable auditing on routes wherein the destination NAP has Analytics
Note behavior selected in its profile.
 Activation of the audit occurs when the audit options are saved; the start and end time will be displayed
on the audit interface. At any time, the audit can be ended using the same configuration interface.

The audit end time is the time limit to start recording calls, thus, when this time limit is
Note reached, no new recordings will start. Recordings in progress will continue until the
end of the classifier and can be stored after the audit end time.

To access audit audio, you must access the KMG through your "FTP" service with the same username
and password valid for Web access in the audio/audit folder. Within this folder there are other folders, representing
the possible classification results and error conditions, as follows:

● Voice Mail: Voice Mail detection; depends on the registered patterns.

● Interception Announcement: Interception announcement, detected by patterns. "For instance: this
number has changed".
● Interception Signaling: Reported upon the detection of the answering signaling within a short period of
time.
● Forward Announcement: Forwarding message, also detected by patterns. A classic example is Nextel's
announcement: "Please hold, your message is being forwarded."
● Unknown Announcement: Unknown ad. It is reported upon answering, when there was a voice before
that, but no pattern was recognized.
● Portability Identification: A beep informing that both the destination and origin numbers are from the
same carrier.
● Ringback Signal: Ring tone, used to differentiate the ringback tone (usually songs chosen by users) from
unknown announcements.
● Short Human Answer: Normal human response, for example: “Hello".
● Long Human Answer: Business human response, for example: "Secretary of Environmental and Social
Studies, Philomena, good morning."
● Fax: Fax equipment answering
● No Speech: Silence after answering. Its detection depends on the configuration of the silence time-out
interval (SilenceTime-out).
● Incomplete: Incomplete audios. This is a generic type of classification used to store the audio when a
call was ended, but there has not yet been a conclusion of the classification feature, for example:
unanswered calls.
● .incoming: Temporary folder. Never add, remove, or modify files in it. Some FTP clients may not have
this folder.

Never remove or modify audit folders. Audit files are never removed by the KMG. Removing
old files is the responsibility of the system administrator. Keeping old files which take up
Attention storage drive space may result in unwanted gateway behavior.

10 Management
In this menu the options for managing the gateway are displayed. In the upper menu there are some
buttons with specific functions.

10. Monitoring KMG through SNMP

The KMG has an SNMP agent that offers help on the state of the equipment, including: Gateway
services, telephony modules (devices), signaling and state of telephony channels, in addition to sending of some
commands. Download of the MIB of the KMG is performed in the unit’s Web interface. MIB is a file that contains a
data base of information on KMG objects that can be consulted.

1. Access the "Management" menu.

2. Click on the "Download gateway MIB" button.

Beginning with version 4.1.42.0, the download of two files is performed: KHOMP-MIB
Note and KHOMP-KMG-MIB.
 More information can be found at the "KQueryServer Manual".
The MIB developed for KHOMP products, in the SMI RFC1065 RFC-1065 format, can be downloaded in
the "Download gateway MIB" option in the Management menu of the Web interface.

10. Gateway information

This button provides all of the basic information about the equipment, model, serial number of the
equipment, Hardware ID that is the unique identification of the equipment, in addition to the version of the packet
installed in the equipment.

10. Remote Terminal - CLI

"Management" → "Remote Terminal"

A new command line interface is made available via SSH. New options can be consulted through the
terminal’s help command. To access it, click on the "Remote Terminal" button, read the instruction displayed in the
Interface and download the private key.

10. Linux Systems

1. In the terminal, browse until you find the directory from which the encryption key was downloaded.
2. Enter the command: chmod 400 kmgXXXXX.khomp.com.br.key, where XXXXX is the serial number of
your device.
3. Enter the command: ssh -i kmgXXXXX.khomp.com.br.key shell@device_IP.
4. Enter the user name.
5. Enter the password.

Notes:
● The login credentials are the same as those used to access the Web interface of the KMG.

10. Turning off the gateway

● Off - Turns the gateway off.
After this process you will need to physically press the on/off button on the device.

10. Restarting the gateway

● Restart - Restart the gateway.
Disables all connections and applies any settings that have been saved and have not yet been applied.

10. Managing system users

Administration → System Users

New users with different Web interface access privileges can be configured. Privileges allow for access
to the four gateway menus: Configuration, Monitoring, Diagnosis, and Administration. For the monitoring menu, you
can specify whether the user can have full or read-only access. These users may access KMG both through the
Web interface and FTP.
 10. Changing a user password
1. In the menu "Administration", click on the icon " " for the user whose password will be changed.
2. Enter the new password in the "New Password" and "Reenter password" fields.
3. Click on the "Save" button.

The new password will be requested when next accessing the KMG.

10. Creating a new user account

1. In the menu "Administration", click on the button "New User".
2. In the field "User" enter the access login. Finally, enter the user password.
3. Define access permissions for this user. It is possible to grant access to the menus: Configuration,
Monitoring, Diagnosis, and Administration. You can grant the following permissions:
● Full: Total permission. The user can visualize and configure the menu options.
● Read: The user can only view the menu options. This permission is only displayed for the
Monitoring menu.
● None: The user will not have access to this section. The menu will not be displayed for this
user.

10. Editing and deleting user accounts

Editing of users consists solely of changing passwords and access permissions for the KMG. Editing is
performed in the panel "System Users".

To remove a user, click on X next to the change password icon.

10. Accessing KMG via FTP

The KMG may also be accessed through the FTP protocol using the same IP address defined for
accessing the Web interface. Access to KMG via FTP is limited to one session at a time.

The following describes actions that can be taken and files that can be accessed through the FTP:

● CDR files: All CDR records are available for download in the /cdr directory. These files should be
removed manually to free up space.
● KMG Log Files: All of the current Log files are available in the /log directory. Older Log files are stored in
the subdirectory /log/old.

These files are managed by the KMG. Older files are removed automatically to free up space on the disc.

● Update files: KMG update files are in the firmware folder. Upload and management of theses files can be
performed both via FTP and via Web interface.
● Classification audit recordings: Recordings from the classification audit will be available in the
audio/audit directory. The files in this folder are not removed automatically and their maintenance is the
user's responsibility.
● Files for playback in the classification list: The files that you wish to play using the Play audio and hang
up feature of the call classifier must be placed in the audio directory.
● Network packet capture files: The network packet capture files will be generated in the capture directory.

● All users can access the FTP.

Note ● All users have access to KMG via FTP protocol.
 10. Licenses
The licenses installed on this device are displayed.
To install a new license, upload it by selecting the file in the Send new license box and click on the
Submit button; the new license will be installed.

It will be necessary to restart the equipment to activate new E1 Links.

Note

10. Provisioning
Provisioning allows for saving the settings applied to the KMG on file. This way you can keep a backup
of the settings and restore when necessary. This file can also be applied to another KMG gateway.

10. Downloading configurations

1. In the "Management" → "Provisioning" menu, click on the "Download the settings backup" button.
2. In the window that will be displayed, select the location where the file will be saved.

A single packaged, compacted file will be generated and exported in the format "tar.gz". You must use
this format for restoring settings.
The file name contains the KMG serial number and the date it was created.

The settings file should not be changed. An improper change could render the file unusable.
Attention

Restore configurations

3. In the "Management" → "Provisioning" menu, click on the "Choose file" button, in the "Restore
configurations" option.
4. Click on the "Send" button. After performing this step, the KMG must be rebooted
5. Restart the gateway through the "Management" → "Restart the gateway" menu.

The IP address of KMG network interfaces is not changed. This way you can maintain
Note IP access after restoring configurations.

10. Applying a provisioning file to another KMG

It is possible to use a provisioning file in another KMG that is compatible, that is, one that contains the
same telephony modules. However it is necessary to meet some criteria. When using only VoIP (SBC), the
procedure is simpler. When KMG has telephony interfaces (TDM), some requirements must be observed.

10. Requirements
● The KMG model in which the settings file will be applied must support the same number of
simultaneous calls or more. Therefore, it is not possible to restore the configurations of a KMG 3200,
which supports up to 2,000 calls, onto a KMG 2000, which supports up to 240 calls. But it is possible to
do otherwise.
● If there are external telephony modules, they must be arranged in the same order and the same quantity
as the gateway in operation. Otherwise, the device structure will be incompatible and the backup will not
be supported.
 10. Procedure
1. Ensure that gateways meet the requirements described above.
2. Check if the KMG that will have the provisioning file, already has the licenses applied. Check the
"Management" → "Licenses" menu.
3. Load the configuration file as described in the "Restore configurations" section.
4. Restart the machine to validate the settings.

10. Update
This section allows for managing KMG version update files.
All available packets are displayed. The "Remove" and "Update" options are displayed for each file. By
clicking on the "Remove" button, the update file will be removed from the system. By clicking on the Update button,
the file will be installed on the system and the system will be automatically restarted after the installation is finished.
To upload a new update file, click on the "Select File" button, choose the file to be uploaded, shown at
the bottom of the file list. Click on "Submit" and wait for the upload to end.

10. Serial port

The serial port is available in KMG 3200 model only and configured for a 115200 speed and a vt100
terminal type; a null modem cable must be used for connection -- http://en.wikipedia.org/wiki/Null_modem -- and
any terminal emulator (minicom, PuTTY, screen, Hyper Terminal, etc.).

Example of access using minicom: $ minicom -b 115200 -D /dev/ttyS0

10. Troubleshooting
My E1 links won’t align

● Common physical problems:

○ Perform tests to verify the quality of the cables.
○ Confirm that the TX/RX pins are connected as indicated in the Installation section of this
guide.
○ "No grounding". All the devices included in the telephony system must necessarily be
grounded and on the very same ground (KMG, PBX, modem, UPS, E1/T1 cable, etc.). Check
the following items:
○ The cable plug is oxidized or has a connection problem.
○ Coaxial cable is broken and has a poor contact.
○ Check if the connectors on the E1/T1 board are in good working condition: perform a
loopback test between the KMG E1/T1 links by crossing the TX and RX pins. If the
parameters are correct, the LED in E1/T1 link configuration will flash in green.
○ External interference. The cable is coated with a metal mesh, and therefore is subject to
electromagnetic interference. It should not be placed within areas of high electromagnetic
interference (high voltage lines, air conditioner, refrigerators, etc).
○ Check the modem, PBX or whatever is connected with coaxial cable onto the E1/T1 of the
KMG with impedance of 75 ohms. The modem connected to this E1/T1 must have the
same impedance, otherwise synchronization problems may occur. Some modems have RJ
45 connectors, and the impedance of twisted-pair cables is equal to 120 ohms. If there is no
impedance selection switch, use an impedance converter (Balun).

Common configuration problems

● Make sure the device to which KMG is connected has the same protocol settings. Remember that when
using ISDN, one end must be set as "ISDN Network" and the other end as "ISDN User".
● The KMG may perform or receive synchronization through the E1/T1 link. If the E1/T1 link is connected
to a public central hub, this option must be enabled for it to "receive" synchronization. If it is connected
to a PBX system, this option should usually be set as to "generate" synchronization.
● The CRC4 checker is enabled in the KMG configuration. If the device connected to the E1/T1 link is set
as CRC4-enabled, then this option must also be enabled in the KMG settings; otherwise it will not be
enabled.
The E1/T1 link is aligned, but the ISDN channels are down

● Verify the Network and the User side of each node.

I have no access to the device because the network interface is not responding

● Check the network cable.

● Check for any IP conflicts in the network.

Call audio is muted

● Check if the TX/RX pin of an E1 is incorrectly connected to the TX/RX pin of another E1.
● Check if KMG is configured with the same Codecs as those of the SIP Server trunk.
● Check if the RTP ports are the same as those of the SIP Server trunk.

Calls are not being routed

● Check if it is necessary to use the packet IP as a reply address.

● Check the need for using the URI user-part as a B number.
● Check if the KMG and the SIP Server are using the same port for SIP signaling.
● Check for inconsistencies in the A number and B number regex.

I forgot the IP address configured in KMG

● The KMG sends in an Ethernet broadcast network packet all network addresses configured on all
network interfaces.
● The packet uses Ethernet Type 0xF00D, which can be used to filter a traffic capture. We recommend the
free software Wireshark to perform the packet capture.
● There is also a verification if at least one network interface is configured. If none is configured within five
minutes of the KMG being booted, then the first one will be configured with address 10.10.10.10 and
network mask 255.0.0.0.
● Depending on the model, the first one may be eth0 or eth1.

Obtaining access to documents

You can find the manual and other documents on our website at www.khomp.com. To register
and access our documentation, please follow these steps:

For unregistered users:

1. At the Khomp website, go to "Technical Support" → "Restricted area".

2. Click on "Register".
3. Choose the profile that best describes you.
4. Register your e-mail address. You must use a corporate e-mail.
5. Fill out the form that was sent to your e-mail. If you do not receive it in your inbox, please check
your spam box.
6. Follow the steps described below to log in to the restricted area.

For registered users:

1. Go to "Technical Support" → "Restricted area".

2. Log in using your registered e-mail address and password.
3. Go to the Documents option. You will be redirected to the Khomp Wiki.

You can also contact our technical support team by e-mail at [email protected] or by
phone at +55 (48) 3722-2930.

Rua Joe Collaço, 253 - Florianópolis, SC

+55 (48) 3722.2930
[email protected]