GFI Kerio Operator Administrator guide
Below you will find brief information for Kerio Operator.
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ADMINISTRATOR GUIDE
Find out how to configure Kerio Operator in different environments and how to set up advanced features.
The information and content in this document is provided for informational purposes only and is provided "as is" with no warranties of any kind, either express or implied, including without limitation any warranties of merchantability, fitness for a particular purpose, and non-infringement. GFI Software disclaims and in no event shall be liable for any losses or damages of any kind, including any consequential or incidental damages in connection with the furnishing, performance or use of this document. The information is obtained from publicly available sources. Though reasonable effort has been made to ensure the accuracy of the data provided, GFI makes no warranty, promise or guarantee about the completeness, accuracy, recency or adequacy of information contained in this document and is not responsible for misprints, out-of-date information, or errors. GFI reserves the right to revise or update its products, software or documentation without notice. You must take full responsibility for your use and application of any GFI product or service.
No part of this documentation may be reproduced in any form by any means without prior written authorization of GFI
Software.
If you believe there are any factual errors in this document, please contact us and we will review your concerns as soon as practical.
GFI and Kerio Operator are trademarks or registered trademarks of GFI Software or its affiliates in the US and other countries. Any other trademarks contained herein are the property of their respective owners.
Kerio Operator is copyright of Kerio. - 1999-2018 Kerio. All rights reserved.
Document Version: 2.5.4
Last updated (month/day/year): 02/03/2018
Contents
1 Introduction to Kerio Operator
2.1 System requirements for Kerio Operator
2.2.2 Kerio Operator Software Appliance
2.2.3 Kerio Operator VMware Appliance
2.3.2 Kerio Operator Box 1000/3000 Series
2.3.3 Setting Up Kerio Operator Box 1220 and 3230
2.3.4 Connecting to Kerio hardware appliances with a serial console
2.4 Logging into Kerio Operator Administration
2.5 Licenses and registrations
2.5.1 Why should you register the trial version?
2.5.2 Registering full version
2.5.3 Registering via a web browser
2.5.4 How do I apply renewals or add-ons to my Kerio product?
2.6.1 Manually uploading a binary image file
2.6.2 Upgrading from versions 1.2.0 and newer
2.6.3 Upgrading from versions 1.1.3 and older
2.7.1 Connecting to VoIP service providers
2.7.2 Displaying the caller number when transferring and redirecting calls
2.7.3 Configuring Kerio Operator with NexVortex
2.7.4 Connecting Kerio Operator to CenturyLink
2.7.5 Connecting Kerio Operator to Deutsche Telekom
2.7.6 Connecting Kerio Operator to Easybell
2.7.7 Connecting Kerio Operator to Net2Phone
2.7.8 Connecting Kerio Operator to NEXCO Networks
2.7.9 Connecting Kerio Operator to QSC
2.7.10 Connecting Kerio Operator to Sipgate.co.uk
2.7.11 Connecting Kerio Operator to Sipgate Deutschland
2.7.12 Connecting Kerio Operator to SIP.US and SIPTRUNK.COM
2.7.13 Connecting Kerio Operator to TelePacific
2.7.14 Connecting Kerio Operator to Teliax
2.7.15 Connecting Kerio Operator to Vitelity
2.7.16 How to configure Kerio Operator to connect to 802.cz
2.7.17 How to configure Kerio Operator to connect to ActiveNetwork
2.7.18 How to configure Kerio Operator to connect to Bandwidth.com
2.7.19 How to configure Kerio Operator to connect to Breezz (NL)
2.7.20 How to configure Kerio Operator to connect to DevopSys
2.7.21 How to configure Kerio Operator to connect to Exetel
2.7.22 How to configure Kerio Operator to connect to fayn.cz
2.7.23 How to configure Kerio Operator to connect to ha-vel.cz
2.7.24 How to configure Kerio Operator to connect to isphone
2.7.25 How to configure Kerio Operator to connect to Megapath
2.7.26 How to configure Kerio Operator to connect to MultiVoice
2.7.27 How to configure Kerio Operator to connect to netphone.cz
2.7.28 How to configure Kerio Operator to connect to OrbTalk
2.7.29 How to configure Kerio Operator to connect to plusTEL in Denmark
2.7.30 How to configure Kerio Operator to connect to sipgate.com
2.7.31 How to configure Kerio Operator to connect to Telephonic Canada
2.7.32 How to configure Kerio Operator to connect to Voicepulse.com
2.7.33 How to configure Kerio Operator to connect to VOIP-Unlimited
2.7.34 How to configure Kerio Operator to connect to VoipVoice
2.7.35 How to configure Kerio Operator to connect to Xphone.cz
2.7.36 How to connect Kerio Operator to Skype Connect
2.8.1 Configuring Kerio Operator and Cisco SPA8800 for calls over an analog telephone line
2.8.2 Configuring Kerio Operator and Grandstream GXW 4104/4108 for calls over analog telephone lines
2.8.4 Configuring Kerio Operator and Grandstream HT503 for calls over analog lines
2.8.6 Configuring Kerio Operator and Well/Yeastar NeoGate TB400 for calls between SIP and EuroISDN
2.8.7 Configuring Kerio Operator and Well/Yeastar NeoGate TG200 for calls between SIP and GSM
2.8.8 Configuring Kerio Operator and Yeastar NeoGate TE100 for calls over analog lines
2.8.9 Configuring PRI telephone service through the Digium VoIP Media Gateway
2.8.10 Connection with Linksys SPA3102 analog (FXS/FXO) to SIP gateway
2.9.1 Inspecting Kerio Operator API communication in a web browser
3.1 Hardware phones and devices
3.1.1 Hardware telephone basic usage
3.1.2 Configuring BLF on Polycom phones
3.1.3 Configuring Cisco / Linksys SPA phones to support more than three callers in a conference
3.1.4 Configuring Snom M300/M700 with Kerio Operator
3.1.5 Configuring the Aastra 6755i IP Phone with Kerio Operator
3.1.6 How to configure Busy Lamp Field (BLF) on Cisco SPA500S
3.1.7 How to configure Busy Lamp Field (BLF) on snom phones
3.1.8 How to configure Busy Lamp Field (BLF) on Well phones
3.1.9 Linksys/Cisco SPA: Setting the TFTP address without using the DHCP parameter 66
3.2.1 Saving Kerio Operator configuration to MyKerio
3.2.2 Saving Kerio Operator configuration to FTP or local storage
3.3 CRM integration and desktop dialers
3.3.1 Salesforce integration with Kerio Operator
3.3.2 Using Kerio Operator App for Salesforce
3.3.3 Configuring OutCALL for dialing from the Microsoft Outlook contacts
3.3.4 CRM integration using the AMI
3.4.1 Using Dashboard in Kerio Operator
3.4.2 Monitoring Kerio Operator
3.4.3 Managing logs in Kerio Operator
4.1.1 Configuring automatic phone provisioning
4.1.2 Provisioning of Kerio Operator Softphone for mobile devices
4.1.3 Accessing company contacts through LDAP on provisioned phones
4.1.4 Using provisioning tools
4.1.5 Editing provisioning templates
4.1.6 Displaying your company logo on the provisioned phones
4.1.7 How to configure phone provisioning on Polycom phones
4.1.8 Phone provisioning - wrong detection of CISCO phones
4.1.9 Uploading configuration files to Kerio Operator TFTP server
4.2.3 Configuring multiple registration of an extension
4.3.1 Mapping external and internal numbers
4.3.2 Displaying, hiding and overriding phone numbers
4.3.3 Setting emergency numbers
4.3.4 Using number transformation
4.3.5 Adding area codes to called numbers
4.4.1 Bandwidth used by the different codecs
4.4.2 Using Opus codec for Kerio Phone
4.4.4 Blocking incoming calls in Kerio Operator
4.4.5 Disabling computer calls for Kerio Phone
4.4.6 Disabling outgoing calls to certain countries or regions
4.4.7 Video calling in Kerio Operator
4.5.2 Configuring music on hold
4.5.4 Configuring and using call parking
4.5.5 Configuring and using conferences
4.5.6 Configuring auto attendant scripts
4.5.7 Setting time conditions in auto attendant scripts
4.5.8 Using the Day/night mode in auto attendant scripts
4.5.10 Configuring call queues
4.6.2 Configuring SSL certificates
4.7.1 Language settings in Kerio Operator
4.7.2 Configuring Built-in DHCP server in Kerio Operator
4.7.3 Configuring parameter 66 in DHCP server in Kerio Control
4.7.4 Configuring server date, time and time zone in Kerio Operator
4.7.5 Configuring standard phone interfaces
4.7.6 Connecting Kerio Operator to directory service
4.7.7 Connecting multiple Kerio Operators
4.7.8 Routing calls between multiple Kerio Operators and the PSTN
4.7.9 Creating and using speed dial
4.7.10 Creating ringing groups
4.7.11 Customization of voice sets
4.7.12 Customizing the Kerio Phone login page
4.7.13 Distinctive ringing support
4.7.14 Fax support in Kerio Operator
4.7.16 Integrating Kerio Connect and Kerio Operator
4.7.17 Setting optional call recording
4.7.18 Setting outgoing calls constraints in Kerio Operator
4.7.20 Using paging groups and services
4.7.21 Configuring Click to Call in Kerio Connect client
5.1.1 Troubleshooting connections to SIP providers
5.1.2 Troubleshooting call quality issues
5.1.3 Browser extensions or add-ons may interfere with Kerio products
5.1.4 Cannot play voicemails or audio files in Safari
5.3.1 Restoring the Kerio Operator default configuration using a USB flash-drive
5.3.2 Restoring the Kerio Operator Box V series default configuration using a USB flash drive
5.3.3 Diagnostic tool for Kerio Operator Box
5.3.4 Diagnostic tool for Kerio Operator Box V series
5.3.5 Recovering password using USB flash-drive for Kerio Operator
5.3.6 Recovering your Kerio Operator Box V series password using a USB flash drive
7.1 Trademarks and registered trademarks
1 Introduction to Kerio Operator
Kerio Operator is a VoIP based phone system that provides enterprise-class voice and video communication capabilities for small and mid-sized businesses globally. Easy to administer and flexible to deploy, as a software appliance, a virtual machine, a hardware appliance, or a cloud solution.
Kerio Operator brings support for high-quality codecs, such as Opus for voice and H.264 for video.
The automatic provisioning feature brings fast setup for various phone systems, such as Cisco, Grandstream, Polycom and
Snom phones.
Its advanced security technologies keep telephone hackers out, prevent misuse and ensure the privacy of your users and those they call. Kerio Operator continually monitors for anomalous behavior, detects and prevents break-in attempts and supports call encryption.
Kerio Operator also includes various call handling features which are used on a daily basis, such as auto attendant scripts, advanced call forwarding, call pickup, Busy Lamp Field (BLF), and many more.
You can stay in control of all your Kerio Operator appliances through Kerio’s centralized web interface -
MyKerio
.
This help system includes technical information about how to deploy, use, configure and troubleshoot Kerio Operator.
Futher reading:
Getting started with Kerio Operator
Logging into Kerio Operator Administration
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2 Getting started
Want to try out Kerio Operator? This topic provides a quick list of actions to help you set it up.
1 Prepare external connectivity
Kerio Operator requires connectivity with a telecommunications service provider (TSP) or an Internet telephony service provider (ITSP) to make and receive external calls. Refer to the supported phone cards and tested SIP providers on the Kerio website.
2 Install Kerio Operator
You can install Kerio Operator as a hardware, software, or virtual appliance. All installation types use a built-in operating system that you manage through the web administration. Refer to the technical specifications
3 Access the Kerio Operator interface
You can administer Kerio Operator directly on the network using a web browser by opening a secure connection to the IP
4 Activate Kerio Operator
When launching the web administration interface for the first time, run through the configuration wizard to activate essential
settings. For more information, refer to Configuration wizard (page 19).
5 Configure Kerio Operator on the network
To communicate on the network, assign network parameters to Kerio Operator from Configuration > Network. Configurable items include NAT, domain name server address for resolving names, IP address, gateway, and subnet for routing to the Internet and local networks.
6 Add extensions and user accounts
To manage calls, you need to
and assign them to your users and phones.
Create
and assign extensions to them.
Connect to a telephone service provider
Telephone service using PRI/BRI, POTS, or Euro-ISDN require physical infrastructure and a hardware interface with specific
Telephone service using a SIP provider requires Internet access. Kerio Operator uses a virtual SIP interface to connect to the SIP
provider. For more information, refer to Connecting to VoIP service providers (page 25).
Deploy user phones
You can manually provision phones through a software interface, or Kerio Operator can automatically provision phones. Click here
You can also use Kerio Operator Softphone app with mobile device or Kerio Phone for desktop operating systems.
You can also use third-party softphones with Kerio Operator. For more information go to http://go.gfi.com/?pageid=operator_ help#cshid=924
2.1 System requirements for Kerio Operator
You can find detailed and always up-to-date
system requirements
for Kerio Operator on our website:
Kerio Operator System Requirements
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2.2 Installing Kerio Operator
2.2.1 Product Editions
Edition
Software
Appliance
VMware
Virtual
Appliance
Kerio
Operator
Box
Description
Kerio Operator Software Appliance is an all-in-one package of Kerio Operator which also includes a special operating system.
Designed to be installed on a computer without an operating system, this edition is distributed as an installation disc.
Software Appliance cannot be installed on a computer with another operating system and it does not allow to install
other applications. For more information, refer to Kerio Operator Software Appliance (page 9).
A virtual appliance designed for use in VMware products.
VMware Virtual Appliance is a Software Appliance edition pre-installed on a virtual host for VMware. The virtual
2.2.2 Kerio Operator Software Appliance
For Kerio Operator system requirements, refer to the Kerio Operator product pages .
You obtain Kerio Operator as a standard ISO image which you need to burn on a CD. Boot from this CD and install the
Kerio Operator operating system. The Kerio Operator application is also installed during the process.
How to connect Kerio Operator Software Appliance to network
After booting the system, a console with the IP address for Kerio Operator is displayed.
If you use a DHCP service on your network, Kerio Operator will be assigned an IP address automatically and will connect to the network. If you do not use or do not wish to use DHCP for Kerio Operator, you have to set the IP address manually.
The current network configuration is displayed (and can be changed) in the Kerio Operator console in section
Network Configuration
. To set a static network address:
1.
Select the
Assign static IP address option in the console menu.
2.
In the network interface on which the PBX should communicate, select the
Assign static IP address option and enter the IP address, subnet mask and IP addresses of gateway and DNS server.
If you know the DNS name of the PBX, you can connect to it and configure it via
.
IMPORTANT
Immediately after you connect Kerio Operator to the network, we recommend to read topic
concerning the security measures
. Meeting security principles for Kerio Operator operation is extremely important. If the PBX is not
protected by a firewall and supporting security rules, your internal telephone extension can be misused which may result in unexpected financial costs.
2.2.3 Kerio Operator VMware Appliance
For supported VMware product versions, check http://www.kerio.com/operator/requirements/
Use an installation package in accordance with the type of your VMware product:
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For products VMware Server, Workstation, Player and Fusion, download the compressed VMX distribution file
(
*.zip
), unpack it and open the file with extension
.vmx
.
You can import a virtual appliance directly to VMware ESX/ESXi from the URL of the OVF file — for example: http://download.kerio.com/dwn/operator/ kerio-operator-appliance-2.3.0-
2500-vmware.ovf
. VMware ESX/ESXi automatically downloads the OVF configuration file and a corresponding disk image (
.vmdk
).
If you import virtual appliance in the OVF format, bear the following specifics in mind:
In the imported virtual appliance, time synchronization between the host and the virtual appliance is disabled.
However, Kerio Operator features a proprietary mechanism for synchronization of time with public Internet time servers.
Therefore, it is not necessary to enable synchronization with the host.
Tasks for shutdown or restart of the virtual machine will be set to default values after the import. These values can be set to hard shutdown or hard reset. However, this may cause a loss of data on the virtual appliance. Kerio Operator
VMware Virtual Appliance supports so called Soft Power Operations which allow to shut down or restart hosted operating system properly. Therefore, it is recommended to set shutdown or restart of the hosted operating system as the value.
For more information, refer to How to connect Kerio Operator Software Appliance to network (page 9).
2.2.4 Kerio Operator Box
For currently supported Kerio Operator Box configurations, refer to the Kerio Operator product pages .
For detailed information on connecting the device into the network, see the
Kerio Operator Box 1000/3000 Series
and
installation guides.
How to connect to the hardware box from the network
Upon the first start, the appliance has a static IP address set to
10.10.10.1
on ethernet port 1. There are two ways to change the configuration:
In the console — use an Ethernet cable to connect to the console. In the console menu, select the
Network Configuration
option and change the configuration.
In the
in section
System
. To connect to Kerio Operator, set the following TCP/IP parameters on your computer: a.
IP address:
10.10.10.2
b.
Subnet mask:
255.255.255.0
To shut down the appliance:
1.
Connect to Kerio Operator via the console and select the
Shutdown command.
2.
Kerio Operator series 1000 will shut down. Kerio Operator series 3000 will stop the server, however, the physical appliance stays switched on. Wait until you are not able to connect to Kerio Operator via Kerio Operator administration and turn the appliance off using the
pwr
button on the appliance.
2.3 Hardware appliances
This section describes deployment and configuration for hardware appliances.
2.3.1 Kerio Operator Box V300 11
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2.3.2 Kerio Operator Box 1000/3000 Series
2.3.3 Setting Up Kerio Operator Box 1220 and 3230
2.3.4 Connecting to Kerio hardware appliances with a serial console
2.3.1 Kerio Operator Box V300
Learn how to safely install and implement Kerio Operator Box V series PBX appliances.
General Safety Instructions
During installation follow these security instructions:
The appliance should be placed on a flat surface.
Do not attempt to open or disassemble the appliance for any reason.
Strictly follow the installation instructions.
Do not place the appliance near a heat source.
Place the appliance in a ventilated space, making sure that the appliance fans and vents are unobstructed at all times.
Do not expose the appliance to liquids of any kind. In the event of liquid intrusion, unplug the appliance immediately.
Verify that the voltage and frequency of the power socket matches the values printed on the power adapter before plugging in the appliance. Use only the power adapter supplied with the appliance.
Do not place any items on top of the power cable; keep the power cable away from walkways or other areas where it could pose a tripping hazard.
Appliance Description
Kerio Operator Box V300 is a Sub-1U table mountable appliance.
Feature Description
Serial port
USB ports
Used for connecting to a console with a serial cable
Input for USB devices
Ethernet network ports Used for connecting to the Internet and the LAN with an Ethernet cable
Kerio Operator Box Installation and Configuration
Once a suitable place has been located for the appliance and it has been plugged into a power outlet according to the safety instructions, it is time to connect it to the network and configure settings.
1.
Connect Ethernet port number 1 to the network using an Ethernet cable.
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NOTE
Alternatively, you can use port number 2 which includes a DHCP client.
2.
Turn on the appliance.
3.
On the computer you want to use for the Kerio Operator configuration, set
IP address
to
10.10.10.2
and
Subnet mask
to
255.255.255.0
. Setting the default gateway and DNS servers is not necessary for the Kerio Operator configuration.
4.
The Kerio Operator PBX is configured through the Kerio Web Administration interface. Open a web browser and connect to the Kerio Control Administration web interface using the
https://10.10.10.1/admin
URL.
5.
Ignore the SSL certificate warning.
6.
Follow the instructions provided by the wizard and configure the appliance.
NOTE
For troubleshooting purposes, you can use the serial port to connect the console to the device.
Additional Information
For further assistance with configuration please refer to additional documentation at: https://manuals.gfi.com/en/kerio/operator/content/home.htm
For online and community based support resources please visit: http://www.kerio.com/support
2.3.2 Kerio Operator Box 1000/3000 Series
Learn how to safely install and implement Kerio Operator Box 1000 and 3000 Series PBX appliances.
General Safety Instructions
During installation follow these security instructions:
The appliance should be placed on a flat surface or securely mounted horizontally in rack enclosure.
Do not attempt to open or disassemble the appliance for any reason.
Strictly follow the installation instructions (see section 4).
Do not place the appliance near a heat source.
Place the appliance in a ventilated space, making sure that the appliance fans and vents are unobstructed at all times.
Do not expose the appliance to liquids of any kind. In the event of liquid intrusion, unplug the appliance immediately.
Verify that the voltage and frequency of the power socket matches the values printed on the power adapter before plugging in the appliance. Use only the power adapter supplied with the appliance.
Do not place any items on top of the power cable; keep the power cable away from walkways or other areas where it could pose a tripping hazard.
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Device Description
Kerio Operator Box types:
Kerio Operator 1000 Series — Sub 1U table mountable appliance (see figure 1).
Kerio Operator 3000 Series — 1U rack mountable appliance (see figure 2).
Screenshot 1: Figure 1 Kerio Operator Box 1220
Screenshot 2: Figure 2 Kerio Operator Box 3230
Feature Description
Serial port
USB ports
Used for connecting to a console with a serial cable
Input for USB devices
Ethernet network ports Used for connecting to the Internet and the LAN with an Ethernet cable
Kerio Operator Box Installation and Configuration
Once a suitable place has been located for the appliance and it has been plugged into a power outlet according to the safety instructions, it is time to connect it to the network and configure settings.
1.
Connect Ethernet port number 1 to the network using an Ethernet cable.
NOTE
Alternatively, you can use port number 2 which includes a DHCP client.
2.
Power the device with the power switch. For 3000 series, the power switch is located in the rear of the device.
3.
On the computer you want to use for the Kerio Operator configuration, set
IP address
to
10.10.10.2
and
Subnet mask
to
255.255.255.0
. Setting the default gateway and DNS servers is not necessary for the Kerio Operator configuration.
4.
The Kerio Operator PBX is configured through the Kerio Web Administration interface. Open a web browser and connect to the Kerio Control Administration web interface using the
https://10.10.10.1/admin
URL.
5.
Ignore the SSL certificate warning.
6.
Follow the instructions provided by the wizard and configure the appliance.
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NOTE
Alternatively, you can use the serial port to connect the console to the device. After the server starts, you can get information about the actual network configuration or you can use the console to restart or turn off the appliance.
Set your terminal application in the following mode: 9600, 8, N, 1.
Additional Information
For further assistance with configuration please refer to additional documentation at: https://manuals.gfi.com/en/kerio/operator/content/home.htm
For online and community based support resources please visit: http://www.kerio.com/support
2.3.3 Setting Up Kerio Operator Box 1220 and 3230
There are two Kerio Operator Box models available:
Kerio Operator Box 1220
- A small desktop appliance featuring six Gigabit Ethernet ports.
Kerio Operator Box 3230
- A 1U rack-mount appliance featuring eight Gigabit Ethernet ports.
WARNING
Kerio Operator Box 3230 is intended primarily for server rooms due to noisy performance.
For more information, refer to Kerio Operator Box 1000/3000 Series (page 12).
2.3.4 Connecting to Kerio hardware appliances with a serial console
Connecting to the Kerio Control hardware appliance through a serial console can help you in the following cases:
Broken network access to the hardware appliance due to configuration mistakes or network hardware issues (both from the box and network switch sides)
Direct access to the Linux shell
You need to see the boot sequence from the hardware appliance
Access to BIOS
Setting a communication through a serial console
The connection uses these settings:
Speed: 9600
Data bits: 8
Stop bit: 1
Parity: none
Flow control: none
Accessing BIOS
The connection uses these settings:
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Speed: 115200
Data bits: 8
Stop bit: 1
Parity: none
Flow control: none
Use the instructions for your operating system to create these settings:
Windows
To connect to the hardware appliance, you need a special application such as PuTTY or RealTerm. Here are the steps for
RealTerm:
1.
Install RealTerm on your computer.
2.
Attach the serial cable to the hardware appliance and to your PC.
3.
Run RealTerm.
4.
On the
Display
tab, select
ANSI
.
5.
Click the
Port
tab and make the following selections there:
Baud: 9600
Parity: None
Data Bits: 8
Stop Bits: 1
Hardware Flow Control: None
6.
Click
Change
.
Before logging on to your hardware device, click
Clear
.
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Now, you can log in to your hardware device as root. Use the admin password for verification.
Linux
To connect to the hardware appliance, you need a special terminal software such as minicom. Here are the steps for minicom:
1.
Install the minicom application.
2.
Type the following command at the shell prompt:
$minicom -s
3.
In the menu, select
Serial port setup
.
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4.
Type
A
.
5.
In the
A
section, type the interface:
TTYS0
. If you use an USB-to-serial adapter, select
USB
instead.
6.
Press
Enter
.
7.
Type
E
.
8.
In the
E
section, type
CQ
.C: 9600 baud, Q: 8 bits, parity: none, stop bit: 1.
9.
Press
Enter
.
10.
Type
F
and set it to
No
.
11.
Press
Enter
to save the configuration.
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12.
Return to the main menu.
13.
Select
Exit
.
Now, you can log in to your hardware device as root. Use the admin password for verification.
OS X
To connect to the hardware appliance, you need:
USB to Serial adapter with the FTDI chipset directly supported by OS X.
Special terminal software such as CoolTerm.
Here are the steps for CoolTerm:
1.
Put the serial cable to the hardware appliance and also to your Mac with the USB to Serial adapter.
2.
Open CoolTerm.
3.
In the
Serial Port
section, select the USB adapter as port.
4.
Baudrate: 9600.
5.
Data Bits: 8.
6.
Parity: none.
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7.
Stop Bits: 1.
8.
Flow Control: no selection.
9.
Click
Connect
.
Now you can log in to your hardware device as root. Use the admin password for verification.
2.4 Logging into Kerio Operator Administration
We recommend to use the supported browsers to connect to Kerio Operator Administration. For the list of the browsers, refer to the Kerio Operator product pages
.
Kerio Operator Administration is currently localized into several languages. Select yours in the top right corner of the interface. The default language is set according to your browser language settings.
2.4.1 How to login
Before you login for the first time, make sure you have:
DNS name of the server with Kerio Operator.
Supported browser
To login, enter the DNS name of the computer with Kerio Operator: kerio.operator.name/admin
Administration runs solely via the HTTPS protocol on port 4021. The address is automatically redirected to: https://kerio.operator.name:4021/admin
NOTE
If the PBX is located behind firewall, HTTPS on port 4021 must be enabled.
If the URL is entered correctly, your browser displays a warning about a SSL certificate. After the installation, Kerio
Operator creates a certificate which is not signed by a trusted certificate authority — it is a self-signed certificate (for more information, read topic about the
SSL certificates ). Since you know the certificate can be trusted, you can add the
security exception and continue to a login page.
Configuration wizard
When you connect to the PBX for the first time, a configuration wizard to do the necessary configuration. Here are those settings:
1.
Set the configuration wizard language.
2.
Accept the Kerio Operator license agreement.
3.
Set a password for the administration account (be sure to remember the password, you will need it to login to the
PBX).
NOTE
This admin password is synchronized with password of user root in the operating system where Kerio Operator is installed (Kerio OS).
4.
Set the time zone of Kerio Operator (requires a restart of the PBX).
5.
Set the PBX language for communication with you and other users (warnings, auto attendant scripts, voicemail, etc.).
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6.
Configure the first extension number. If you use phone provisioning, extensions will be created automatically beginning with the number you enter here.
After successful configuration, the login page is displayed. Enter the username and password you created earlier.
To change the password, use the following steps:
1.
Login to Kerio Operator using the HTTPS protocol (e.g.
https://operator.company.com/admin
)
2.
Open the
Configuration > Users
section.
3.
In the user list, select the administrator account you are logged in with and double-click on it.
4.
Change the password on tab
General
.
2.5 Licenses and registrations
You can register the product from the welcome page of the administration interface which is displayed after each login.
WARNING
If Kerio Operator is protected by a firewall, it is necessary to allow outgoing HTTPS traffic for Kerio Operator at port
443. Unless HTTPS traffic is allowed, Kerio Operator cannot use the port to connect to the Kerio Technologies registration server.
When installed, the product can be registered as trial or as a full version.
2.5.1 Why should you register the trial version?
The trial version is intended to allow the customer to become familiar with the product's features and configuration.
Once you register the trial version, you will be provided free Kerio Technologies technical support during the entire trial period (up to 30 days).
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The trial version can be registered by clicking
Become a registered trial user
from the
Dashboard
(see the screenshot above). In the dialog box that appears, set the following parameters:
1.
Enter security code (CAPTCHA) from the image.
2.
Enter information about your company and agree with the privacy policy terms.
3.
Choose how many computers do you have in your company and how you learned of Kerio Operator.
Now, a special identification code called Trial ID gets generated. This ID is later required for contacting the technical support. After a successful registration, Trial ID can be found in the license information in the administration interface.
NOTE
Once you purchase the product, your Trial ID will become your license number (it will not change).
2.5.2 Registering full version
If your trial version is registered, the license key ( licence.key
file) is automatically imported to your product within
24 hours from your purchase. The Trial ID you entered in your product upon registration will be activated as a standard license number.
If you haven’t registered your trial version:
1.
Open the administration interface.
2.
Click
Register product with a purchased license number
on Dashboard.
3.
In the first step of the registration, enter the license number and enter the security code from the image.
NOTE
The code is not case-sensitive.
4.
Click
Next
to make Kerio Operator establish a connection to the registration server and check validity of the number entered. If the number is invalid, the registration cannot be completed.
5.
Type the registration information about the company the product is registered to.
6.
Kerio Operator connects to the registration server, checks whether the data inserted is correct and downloads automatically the license key (digital certificate).
7.
Click
Finish
to close the wizard.
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Manual import the license key
If you need to import a license key manually (for example from a backup), use the following steps:
1.
Prepare the license key.
2.
Log in to Kerio Operator administration.
3.
Click
Install license
on
Dashboard
.
4.
In the
Install License
dialog, click
Browse
.
5.
In the
Open
dialog, find the file
.key
with the license key and click
Open
.
6.
In the
Install License
dialog, click
OK
.
7.
Check the result in the
License
tile on
Dashboard
.
Kerio Operator installs the licence key.
2.5.3 Registering via a web browser
You can also register Kerio Operator via web browser.
1.
Go to https://secure.kerio.com/reg/
2.
Register using your purchased license number.
3.
By registering, you will receive a license key which must be
.
NOTE
The trial version of Kerio Operator cannot be registered via the website.
2.5.4 How do I apply renewals or add-ons to my Kerio product?
When you purchase renewals or add-ons for a Kerio Product, License changes are applied automatically by the product within 24 hours.
You can also force an immediate update from the administration dashboard using the
update registration info
" link in the
License Details
tile.
2.6 Upgrading Kerio Operator
Choose your current Kerio Operator version for notes and instructions on how to upgrade to the latest version while retaining all settings:
Kerio Operator 1.2.0 and newer
Kerio Operator 1.1.3 and older
2.6.1 Manually uploading a binary image file
This procedure might be useful for the following situations: downgrade of Kerio Operator upgrade to a custom version (e.g. beta version)
If you have an upgrade image file, you can upload it manually:
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1.
In the administration interface, go to section
Advanced Options > tab Update Checker
.
2.
Click the
Upload binary image
button.
3.
Select the upgrade image file ( kerio-Operator-upgrade.img
).
4.
Click the
Open
button. Wait for uploading the file.
5.
Click the
Upgrade
button. Wait for the upgrade and restart of Kerio Operator.
When the restart is finished, your Kerio Operator is up-to-date.
2.6.2 Upgrading from versions 1.2.0 and newer
Learn how to upgrade Kerio Operator to the latest version while retaining all settings.
Important notes when upgrading
An active and valid Software Maintenance is required to upgrade to new versions of Kerio Operator and its components as soon as they are available.
Backup the Kerio Operator configuration before upgrade.
For more information, refer to Saving Kerio Operator configuration to MyKerio (page 144).
Check that the server meets the latest
system and hardware requirements
.
Kerio Operator requires restarts during upgrade. Perform the upgrade when there is no traffic on the server or when it is least impacting on the business operation.
Upgrade procedure
1.
From the administration console, go to
Advanced Options > Update Checker
.
2.
Select the
Periodically check for new versions
option, so Kerio Operator checks for new updates every 24 hours.
3.
If you want to download new versions automatically, select
Download new versions automatically
. If you want to get also beta versions of the product, select
Check also for beta versions
.
4.
Click
Apply
.
5.
When Kerio Operator finds a new version, click
Upgrade
to install it. Click
Yes
to confirm.
6.
Kerio Operator then upgrades to the latest version and restarts automatically when done.
2.6.3 Upgrading from versions 1.1.3 and older
Learn how to upgrade Kerio Operator to the latest version while retaining all settings.
Important notes when upgrading
An active and valid Software Maintenance is required to upgrade to new versions of Kerio Operator and its components as soon as they are available.
Backup the Kerio Operator configuration before upgrade.
For more information, refer to Saving Kerio Operator configuration to MyKerio (page 144).
Check that the server meets the latest
system and hardware requirements
.
Kerio Operator requires restarts during upgrade. Perform the upgrade when there is no traffic on the server or when it is least impacting on the business operation.
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Upgrade procedure
1.
Increase the file upload limit so Kerio Operator can be upgraded to a newer version. To do this, from the administration console, go to
Advanced Options
and set the maximum value in the
Maximum webserver upload file size
field.
2.
Restart Kerio Operator.
3.
Download
Kerio Operator 1.2.0
.
4.
Go to
Advanced Options > Update Checker
, click the
Upload binary image
button, select the upgrade image file, wait until the file is uploaded and click
Upgrade
.
5.
After Kerio Operator is restarted, go to
Advanced Options >Update Checker
, select the
Periodically check for new versions
option, and click
Apply
.
6.
After Kerio Operator finds the latest version, click
Upgrade
, and click
Yes
to confirm the action. When the upgrade process is done, Kerio Operator restarts automatically.
2.7 Provider setup
This section helps you connect to various SIP providers.
2.7.1 Connecting to VoIP service providers
2.7.2 Displaying the caller number when transferring and redirecting calls
2.7.3 Configuring Kerio Operator with NexVortex
2.7.4 Connecting Kerio Operator to CenturyLink
2.7.5 Connecting Kerio Operator to Deutsche Telekom
2.7.6 Connecting Kerio Operator to Easybell
2.7.7 Connecting Kerio Operator to Net2Phone
2.7.8 Connecting Kerio Operator to NEXCO Networks
2.7.9 Connecting Kerio Operator to QSC
2.7.10 Connecting Kerio Operator to Sipgate.co.uk
2.7.11 Connecting Kerio Operator to Sipgate Deutschland
2.7.12 Connecting Kerio Operator to SIP.US and SIPTRUNK.COM
2.7.13 Connecting Kerio Operator to TelePacific
2.7.14 Connecting Kerio Operator to Teliax
2.7.15 Connecting Kerio Operator to Vitelity
2.7.16 How to configure Kerio Operator to connect to 802.cz
2.7.17 How to configure Kerio Operator to connect to ActiveNetwork
2.7.18 How to configure Kerio Operator to connect to Bandwidth.com
2.7.19 How to configure Kerio Operator to connect to Breezz (NL)
2.7.20 How to configure Kerio Operator to connect to DevopSys
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2.7.21 How to configure Kerio Operator to connect to Exetel
2.7.22 How to configure Kerio Operator to connect to fayn.cz
2.7.23 How to configure Kerio Operator to connect to ha-vel.cz
2.7.24 How to configure Kerio Operator to connect to isphone
2.7.25 How to configure Kerio Operator to connect to Megapath
2.7.26 How to configure Kerio Operator to connect to MultiVoice
2.7.27 How to configure Kerio Operator to connect to netphone.cz
2.7.28 How to configure Kerio Operator to connect to OrbTalk
2.7.29 How to configure Kerio Operator to connect to plusTEL in Denmark
2.7.30 How to configure Kerio Operator to connect to sipgate.com
2.7.31 How to configure Kerio Operator to connect to Telephonic Canada
2.7.32 How to configure Kerio Operator to connect to Voicepulse.com
2.7.33 How to configure Kerio Operator to connect to VOIP-Unlimited
2.7.34 How to configure Kerio Operator to connect to VoipVoice
2.7.35 How to configure Kerio Operator to connect to Xphone.cz
2.7.36 How to connect Kerio Operator to Skype Connect
2.7.1 Connecting to VoIP service providers
NOTE
This information is designed for Kerio Operator 2.4 and newer.s
You can connect Kerio Operator either to your VoIP service provider's SIP server or to a
topic discusses about connecting to a VoIP service provider.
Prerequisites
Before you configure an interface, you need the following information:
Telephone number (or numbers) from your SIP provider.
Domain/hostname of SIP server.
Username and password for authentication.
At least one internal extension defined in Kerio Operator — preferably the extension of an employee who redirects the calls.
Adding an interface
To configure an interface, you must first configure call routing. Once you configure incoming call routing, a configuration wizard configures outgoing call routing automatically.
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1.
In the administration interface, go to
Configuration > Call Routing
and click
Add SIP interface
. This displays the configuration wizard.
2.
Key in a name for the interface (for example, the provider's name). The name must not contain spaces or special characters and must be unique.
3.
Select
New provider
. The configuration differs for settings with
one number or multiple numbers
and for a SIP trunk with an
One or multiple phone numbers
1.
If you acquire one or multiple phone numbers from your provider, key in the numbers in the
New provider > With external number
field. You can:
Separate individual numbers with commas (for example,
555450, 555451, 555452, and so on).
Key in a range using a dash (for example,
555450-555459
).
2.
Click
Next
.
3.
Select an extension that receives all calls from the provider.
4.
Optionally, in the
Prefix to dial out
field, key in a prefix for outgoing calls and click
Next
. Kerio Operator uses this prefix to route calls to your provider's SIP server. This prefix can be the same for other providers.
For more information, refer to Working with prefixes for outgoing calls (page 202).
5.
Key in the domain name or the IP address acquired from your provider and if the server requires authentication, also key in the username and password.
6.
Select
Required to register
(the majority of providers require registration to a SIP server) and click
Next
.
7.
Verify the information in the
Summary
section. If you need to add more information from your provider (for example, outbound proxy, inbound proxy, registrar, and so on), select the
Edit details of the created interface
option.
For more information, refer to Configuring additional SIP details (page 27).
8.
Click
Finish
.
9.
Optionally, double-click the interface and enable the
Send keep-alive requests every 20 seconds
option.
WARNING
If your SIP provider does not send keep-alive packets, or your firewall or router has short and unchangeable NAT timeout for UDP connections, use this option to keep the UDP session open.
10.
Click
OK
to save your changes.
11.
Create a rewriting rule to correctly map numbers to internal user extensions.
For more information, refer to Mapping external and internal numbers (page 196).
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Interval of numbers
1.
If you acquire a SIP trunk with an interval of phone numbers from your provider, key in x in place of the digits that vary (for example,
555xxx
).
2.
Click
Next
.
3.
Select the extension on which you want Kerio Operator to redirect all calls to unassigned (unused) extensions.
4.
Optionally, in the
Prefix to dial out
field, key in a prefix for outgoing calls. Kerio Operator uses the prefix to route calls to your provider's SIP server. This prefix can be the same for other providers.
For more information, refer to Working with prefixes for outgoing calls (page 202).
5.
Click
Next
.
6.
Key in the domain name or the IP address acquired from your provider. If the server requires authentication, also key in the username and password.
7.
Select the
Required to register
option if the provider requires registration. With large number intervals, some providers do not require registration. Instead they use the IP address of your Kerio Operator. The address must be static and the provider needs to know about any changes that may occur.
8.
Verify the information in the
Summary
section. If you need to add more information from your provider (for example, outbound proxy, inbound proxy, registrar, and so on), select the
Edit SIP details of created interface
option.
For more information, refer to Configuring additional SIP details (page 27).
9.
Click
Finish
.
10.
Create a rewriting rule to correctly map numbers to internal user extensions.
For more information, refer to Mapping external and internal numbers (page 196).
Configuring additional SIP details
To set additional settings in your interface for incoming and outgoing calls:
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
, select a SIP interface and click
Edit
.
2.
On the
SIP Details
tab, you can:
Key in addresses to outbound proxy, inbound proxy and registrar (Kerio Operator uses domain by default).
Change the transport protocol.
Change the
Key in an authentication username (Kerio Operator uses the SIP username by default).
Change outgoing headers.
3.
Click
OK
to save your changes.
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Configuring DTMF method
NOTE
This functionality exists since Kerio Operator 2.4.
For some SIP providers, the default configuration of DTMF detection,
Auto (RFC 2833 / In-band)
, does not work. You must find out the correct method from your SIP provider and configure it manually, as follows:
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select a SIP interface and click
Edit
.
3.
Go to the
SIP Details
tab.
4.
Select the correct
DTMF method
.
5.
Click
OK
.
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Configuring outgoing headers
NOTE
This functionality exists since Kerio Operator 2.4.
For some providers, you must add additional configuration to the SIP headers as provided to you by them.
To configure outgoing headers:
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select a SIP interface and click
Edit
.
3.
Go to the
SIP Details
tab.
4.
Enable the outgoing header (see the list of supported headers below).
5.
Double-click in the
Value
column and key in the header content (see the list of supported variables below).
6.
Click
OK
.
Kerio Operator supports these headers:
From number
P-Preferred-Identity
P-Asserted-Identity
Remote-Party-ID
Diversion
To edit headers, use these variables:
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EXTERNAL_NUMBER shows the external number after number rewriting
ORIGINAL_NUMBER shows the number of the caller
REQUEST_URI requests the information from the header of the forwarded call
DIVERSION_REASON sends the reason of the call forwarding
DOMAIN shows the domain of the interface
Reading the Caller ID from outgoing headers
NOTE
This functionality exists since Kerio Operator 2.4.4.
If your provider does not send the information about calling or called numbers in default headers (
From
for calling number and
Request-Line
for called numbers), you can configure Kerio Operator to read this information from different headers (for example,
P-Asserted-Identity
):
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select a SIP interface and click
Edit
.
3.
Go to the
SIP Details
tab.
4.
For the fields
Read calling number from
and
Read called number from
, select a new header.
5.
Click
OK
to save your settings.
Displaying the caller's number when transferring and redirecting calls
Resolving domain names of SIP providers
Your SIP providers may change their IP address for your registration without prior notice. To avoid inaccessibility, configure Kerio Operator to periodically resolve domain names and renew the registration:
1.
In the administration interface, go to
Call Routing > Interfaces and routing of incoming calls
.
2.
Select a SIP interface and click
Edit
.
3.
Go to the
SIP Details
tab.
4.
Select the
Periodically resolve domain names
option.
5.
Click
OK
.
Kerio Operator now periodically resolves domain names of your SIP provider and renews your registration whenever the
IP address changes.
Mapping of numbers
For more information, refer to Mapping external and internal numbers (page 196).
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2.7.2 Displaying the caller number when transferring and redirecting calls
Kerio Operator enables users to transfer or redirect their calls to another number or device. By default, the other device displays the number assigned to the extension from which the call is forwarded instead of the caller's number.
To solve this issue, enable additional outgoing headers to send the information about the call in them. Ask your provider which outgoing header to use.
Kerio Operator uses the diversion header by default.
NOTE
If your device can read this information, you might see, for example, a different icon or both numbers on your display.
NOTE
This configuration does not affect internal calls.
Configuring the diversion header
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select a SIP interface and click
Edit
.
3.
Go to the
SIP Details
tab.
4.
Under the
Miscellaneous
section, open the drop down list and select
Diversion RFC 5806
. Kerio Operator automatically changes values in the From and Diversion headers.
5.
Click
OK
.
For more information, refer to Configuring outgoing headers (page 29).
Example
In this example:
Meg Regret has the external number
377-338-9010
.
John Smith has external number
555-1111
, which belongs to internal extension
11
.
John configures call forwarding to his cell phone number
555-8888
.
John wants to be able to return forwarded calls directly.
The SIP provider uses a diversion header.
After Meg dials
555-1111
:
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1.
The SIP provider sends the call to Kerio Operator and the call reaches extension
11
.
2.
Kerio Operator redirects the call to
555-8888
.
3.
John Smith calls Meg Regret back. With the diversion header enabled, the call goes directly to Meg Regret. With the diversion header disabled, the call goes back to John's internal extension.
The difference is that, although the call takes the exact same path to the device, the diversion header allows John to read and see the caller's number, and dial back directly to that number.
2.7.3 Configuring Kerio Operator with NexVortex
You can configure Kerio Operator to send and receive calls using a SIP trunk from NexVortex .
Prerequisites
Before starting this procedure, you should have:
The telephone number or numbers assigned to you by NexVortex. Each number will include the international country code without
+ at the beginning. For example,
14085555555
.
Your SIP (PROXY) login credentials provided during the nexVortex account activation.
If you have a firewall, make sure the SIP and RTP ports are properly routed to Kerio Operator.
For more information, refer to Securing Kerio Operator (page 254).
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Configuration
1.
to the administration interface of Kerio Operator.
2.
Go to the Call Routing screen
3.
Click
Add a SIP Interface...
.”
4.
Enter an interface name (e.g. "nexvortex”) and your external number or numbers.
5.
Choose the extension to receive incoming calls
6.
Specify the hostname (nexvortex.com) along with your Username and Password. Enable
Required to register
.
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7.
On the last page of the wizard, enable
Edit details of the created interface
.
8.
In the SIP Details dialog configure the following:
Outbound proxy: nexvortex.com
Inbound proxy: px3.nexvortex.com
, px5.nexvortex.com
, px7.nexvortex.com
Registrar: nexvortex.com
DTMF method:
RFC 2833
Leave all other options with the default settings.
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2.7.4 Connecting Kerio Operator to CenturyLink
NOTE
This topic is meant only for the CenturyLink IQ® SIP Trunk offer. CenturyLink supports only G.729 and G.711 U-law codecs.
Prerequisites
To connect your Kerio Operator to the CenturyLink provider, you need the following information:
Your telephone numbers from CenturyLink. CenturyLink provides two types of numbers:
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Trunk Pilot Number
, which is used as a SIP username.
Trunk or range of external numbers, which are used for managing calls.
The SIP username (
Trunk Pilot Number
) and password (
Trunk Group SIP Password
).
The domain of CenturyLink (voip.centurylink.com).
The outbound and inbound proxies of CenturyLink (
CenturyLink SBC IPv4 Address/Subnet Mask
).
The registrar of CenturyLink (
CenturyLink SBC IPv4 Address/Subnet Mask
).
The authentication username (
Trunk Group SIP ID
).
Configuration
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface.
4.
In the
With external number
field, key in the trunk or range of numbers and click
Next
.
5.
Select an extension to which you want Kerio Operator to redirect all calls to unassigned extensions.
6.
(Optional) In the
Prefix to dial out
field, key in a prefix for outgoing calls.
7.
Click
Next
.
8.
In the
Domain (IP address/hostname)
field, key in voip.centurylink.com
.
9.
In the
Username
field, key in your
Trunk Pilot Number
.
10.
In the
Password
field, key in your
Trunk Group SIP Password
.
11.
Select the
Required to register
option and click
Next
.
12.
Select
Edit details on the created interface
and click
Finish
.
After you finish the configuration, the
Edit External Interface (SIP)
dialog box opens:
1.
In the
Proxies and Registar
section, key in your
CenturyLink SBC IPv4 Address/Subnet Mask
into
Outbound proxy
,
Inbound proxy
, and
Registrar
fields.
2.
Go to the
Miscellaneous
section.
3.
In the
Authentication username
field, key in your
Trunk Group SIP ID
.
4.
In the
Read called number from
field, select
"To" header
.
5.
In the
Outgoing headers
table: Double-click the value for
From number
and key in
ORIGINAL_NUMBER
.
6.
Enable the header.
7.
Double-click the value for
P-Asserted-Identity
and key in
<sip:[email protected]>
, where
Trunk_Pilot_Number represents the number used for the SIP username.
8.
Click
OK
.
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2.7.5 Connecting Kerio Operator to Deutsche Telekom
Prerequisites
To connect your Kerio Operator to Deutsche Telekom, you need the following information:
Your telephone numbers from Deutsche Telekom
The SIP username and the SIP password
The domain of Deutsche Telekom. For example, tel.t-online.de
.
Configuration
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface.
4.
In the
With external number
field, key in your numbers.
NOTE
If you have multiple numbers from Deutsche Telekom, write individual numbers separated by a comma (for example,
555 5501, 555 5502, 555 5503, 555 5504
) or use a dash to define the range of numbers (for example,
555 5501—555 5504
).
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5.
Click
Next
.
6.
Select an extension that receives all calls from the provider by default.
7.
Optionally, In the
Prefix to dial out
field, you can key in a prefix for outgoing calls.
8.
Click
Next
.
9.
In the
Domain (IP address/hostname)
field, key in tel.t-online.de
.
10.
Key in the username and password.
11.
Select the
Required to register
option.
12.
Click
Next
.
13.
Select the
Edit details of the created interface
option and click
Finish
.
After you finish the configuration wizard, the
Edit External Interface (SIP)
dialog box opens:
1.
In the
Proxies and Registrar
section, enable the
Periodically resolve domain names
option.
2.
Click
OK
.
2.7.6 Connecting Kerio Operator to Easybell
Prerequisites
To connect your Kerio Operator to Easybell, you need the following information:
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Your telephone numbers from Easybell.
The SIP username and password.
The domain of Easybell ( sip.easybell.de
).
Configuration
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface.
4.
In the
With external number
field, key in the range of your numbers.
5.
Click
Next
.
6.
Select the extension to which you want Kerio Operator to redirect all calls to unassigned (unused) extensions.
7.
Optionally, in the
Prefix to dial out
field, you can key in a prefix for outgoing calls.
8.
Click
Next
.
9.
In the
Domain (IP address/hostname)
field, key in sip.easybell.de
.
10.
Key in the username and password.
11.
Select the
Required to register
option.
12.
Click
Next
.
13.
Verify the information in the
Summary
section and click
Finish
.
14.
Double-click the created outgoing route for this interface.
15.
In the
Calling number (Caller ID)
section, select
Map extensions to external numbers based on the incoming routing table
and click
OK
.
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2.7.7 Connecting Kerio Operator to Net2Phone
Learn how to configure Kerio Operator with a SIP Trunk to Net2Phone.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before you start the configuration, you need the following information:
The telephone number or numbers assigned to you by Net2Phone. Each number will include the US international country code (without
+ at the beginning). For example,
14085555555
.
Your SIP (PROXY) login credentials provided during the Net2Phone account activation.
Configuration
1.
In the administration interface, go to
Configuration > Call Routing
.
2.
Click
Add SIP Interface
.
3.
In the first screen, key in an interface name (for example,
Net2Phone
), select
New provider
and key in your telephone number. For example,
14085555555
. In case of multiple numbers, use comma separation as noted in the dialog.
4.
Click
Next
.
5.
Choose the extension to receive incoming calls and key in a prefix that will be used for external calls (for example,
9
).
6.
Click
Next
.
7.
In the
Domain (IP address/hostname)
field, specify the hostname and credentials as provided by Net2Phone
( ippbx.net2phone.com
).
8.
Check
Require to register
.
Check the settings by dialing an external phone number.
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2.7.8 Connecting Kerio Operator to NEXCO Networks
NOTE
This information is designed for Kerio Operator 2.3.5 and older.
For more information, refer to Connecting to VoIP service providers (page 25).
You can configure a SIP trunk with NEXCO Networks for dialing to the public telephone network. This topic describes the necessary configuration in Kerio Operator.
Prerequisites
After setting up an account with NEXCO Networks, you should be given the following information from the provider:
The telephone number (or numbers) assigned to you.
Trunk # (sometimes referred to as User Name or User ID).
Password
Server IP or Domain. For example, media1.nexconetworks.net
.
This information is required in the configuration as described below.
Configuration
1.
to the web administration interface of Kerio Operator.
2.
Go to
Configuration > Call Routing
.
3.
Click
Add a SIP Interface.
4.
Enter an interface name (e.g. "NEXCO”).
5.
Choose
New provider
and enter your telephone number (you may need to add a
1 at the beginning of the telephone number). Use a comma to separate multiple phone numbers.
6.
Click
Next.
7.
Choose the extension to receive incoming calls.
8.
Enter a dial out prefix if necessary.
9.
Click
Next.
10.
Specify the hostname provided by NEXCO Networks (
media1.nexconetworks.net
) and the default port
5060.
11.
Specify the
Username
(Trunk #) and
Password
(Password) values as provided by NEXCO Networks.
12.
Enable
User ID differs from the telephone number
and enter your NEXCO Networks Trunk # (the same number that is in the Username field).
13.
Enable the option
Register with registrar.
14.
Click
Finish.
15.
In order to transmit the correct CallerID information on outbound calls you must edit the interface that you’ve just created by doing the following:
Select the interface you just created (e.g. NEXCO) and click
Edit.
Click the
Advanced
tab.
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Check the box
Use SIP user ID in REGISTER request only.
Click
OK
to save.
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2.7.9 Connecting Kerio Operator to QSC
Prerequisites
To connect your Kerio Operator to QSC, you need the following information:
Your telephone numbers from QSC.
The SIP username and the SIP password.
The domain of QSC (sip.qsc.de).
Configuration
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
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3.
Key in a name for the interface.
4.
In the
With external number
field, key in the range of your numbers.
5.
Click
Next
.
6.
Select the extension to which you want Kerio Operator to redirect all calls to unassigned (unused) extensions.
7.
Optionally, in the
Prefix to dial out
field, you can key in a prefix for outgoing calls.
8.
Click
Next
.
9.
In the
Domain (IP address/hostname)
field, key in sip.qsc.de
.
10.
Key in the username and password.
11.
Select the
Required to register
option.
12.
Click
Next
.
13.
Select the
Edit details of the created interface
option and click
Finish
.
After you finish the configuration wizard, the
Edit External Interface (SIP)
dialog box opens:
1.
Go to
SIP Details > Miscellaneous
.
2.
In the
Read called number from
field, select the
"To" header
option.
3.
Click
OK
.
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2.7.10 Connecting Kerio Operator to Sipgate.co.uk
Prerequisites
To connect your Kerio Operator to Sipgate.co.uk, you need the following information:
Your phone numbers from Sipgate.co.uk
WARNING
In the Sipgate.co.uk account, Sipgate.co.uk displays numbers in the UK format. When you configure the numbers in
Kerio Operator, you need to change the prefix
0
to a prefix
44
.
For example, if you have numbers
056 0001 2345 and
056 0001 2346 from Sipgate.co.uk. When you configure the SIP interface in Kerio Operator, you change the prefix and key in numbers
4456 0001 2345 and
4456 0001 2346
.
The SIP-ID of your SIP account. For example,
123456t0
.
The SIP password of your SIP account.
The Registry/Proxy address of your account. For example, sipconnect.sipgate.co.uk
.
Configuration
In the administration interface of Kerio Operator:
1.
Go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface.
4.
In the
With external number
field, key in your numbers with changed prefixes.
For more information, refer to Prerequisites (page 46).
.
5.
Click
Next
.
6.
Select the extension that receives all calls from the provider.
7.
Optionally, in the
Prefix to dial out
field, you can key in a prefix for outgoing calls.
8.
Click
Next
.
9.
In the
Domain (IP address/hostname)
field, key in sipconnect.sipgate.co.uk
.
10.
Key in the username (
123456t0
) and the password.
11.
Select the
Required to register
option and click
Next
.
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12.
Select the
Edit details of the created interface
option and click
Finish
.
Kerio Operator finishes the configuration wizard and the
Edit External Interface (SIP)
dialog box opens:
1.
Go to the
Outgoing headers
table.
2.
In the
From number
field, key in your SIP-ID (
123456t0
)
3.
Optionally, select
P-Preferred Identity
and do not change the default value. If you have more than one phone number from Sipgate.co.uk, use this option to display the external numbers configured in Kerio Operator for your outgoing calls instead of the
Fallback Caller ID
number configured in your Sipgate account.
4.
Click
OK
.
You can now make some test calls to verify the connection to Sipgate.co.uk.
2.7.11 Connecting Kerio Operator to Sipgate Deutschland
Prerequisites
To connect your Kerio Operator to Sipgate Deutschland, you need the following information:
Your telephone numbers from Sipgate Deutschland.
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NOTE
You need to add the country code 49 to the external number.
The SIP username and the SIP password.
The domain of Sipgate Deutschland (sipconnect.sipgate.de).
Configuration
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface.
4.
In the
With external number
field, key in the range of your numbers.
5.
Click
Next
.
6.
Select the extension to which you want Kerio Operator to redirect all calls to unassigned (unused) extensions.
7.
Optionally, in the
Prefix to dial out
field, you can key in a prefix for outgoing calls.
8.
Click
Next
.
9.
In the
Domain (IP address/hostname)
field, key in sipconnect.sipgate.de
.
10.
Key in the username and password.
11.
Select the
Required to register
option.
12.
Click
Next
.
13.
Select the
Edit details of the created interface
option and click
Finish
.
After you finish the configuration wizard, the
Edit External Interface (SIP)
dialog box opens:
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1.
Go to
SIP Details > Miscellaneous > Outgoing headers
.
2.
In the
From number
header, key in your SIP username.
3.
Select the
P-Preferred-Identity
header and click
OK
.
2.7.12 Connecting Kerio Operator to SIP.US and SIPTRUNK.COM
You can configure a SIP trunk with
SIP.US
or
SIPTRUNK.COM
for dialing to the public telephone network. This topic describes the necessary configuration in Kerio Operator.
Prerequisites
Accounts you setup in SIP.US or SIPTRUNK.COM include the following information (available in the SIP.US or
SIPTRUNK.COM control panel) which is required during the configuration with Kerio Operator:
The telephone number or numbers assigned to you.
Trunk #
Password
Configuration
1.
to the web administration interface of Kerio Operator.
2.
Go to
Configuration > Call Routing
.
3.
Click
Add a SIP Interface
.
4.
Assign an interface name.
5.
Choose
New provider
and enter your telephone number (use a comma to separate multiple phone numbers).
6.
Click
Next
.
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7.
Choose the extension to receive incoming calls.
8.
Enter a dial out prefix if necessary.
9.
Click
Next
.
10.
Specify the domain (gw.sip.us for SIP.US) or (gw.siptrunk.com for SIPTRUNK.COM)
11.
Specify the
Username
(Trunk #) and
Password
(Password) values.
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12.
Enable
Required to register
.
13.
Click
Next
.
14.
Review your settings and click
Finish
.
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2.7.13 Connecting Kerio Operator to TelePacific
You can configure a SIP trunk with Telepacific for dialing to the public telephone network. This topic describes the necessary configuration in Kerio Operator.
Prerequisites
To maximize call quality, TelePacific installs a customer-premises equipment (CPE) router at your physical location. It is necessary to properly design your network to support this type of configuration.
Kerio Operator requires at least two network interfaces. One interface connects directly to the CPE router, and the other interface connects to your local area network. The diagram below illustrates the basic structure.
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Configuring TCP/IP parameters in Kerio Operator
Configure the interface connecting to the CPE Router with a private IP address that you assign manually (e.g.,
192.168.10.11/24). Do not assign a gateway to this interface. Configure the interface connecting to your local network with a static, or dynamic IP address that your DHCP server assigns to Kerio Operator. Use this interface to connect to the internet via the local area network.
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1.
to the administration interface
2.
Go to
Configuration > Network
3. Edit the interface connecting to the CPE router and assign the TCP/IP parameters for this private network
4. Edit the interface connecting to the local network and verify that the TCP/IP parameters are valid for the local area network
5.
Click
Apply
Configuration
1.
to the administration interface and go to
Configuration > Call Routing
2.
Click
Add a SIP Interface
and enter an interface name (e.g. "Telepacific”)
3.
Choose
New provider
and enter your telephone number (use a comma to separate multiple phone numbers)
4.
Click
Next
5.
Choose the extension to receive incoming calls and leave the dial out prefix empty
6.
Click
Next.
7.
Specify the IP address of the CPE Router (e.g., 192.168.12.1) and do not change the default port
8.
Uncheck
Required to register
and leave the
Username
and
Password
fields empty
9.
Click
Next
and
Finish
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2.7.14 Connecting Kerio Operator to Teliax
You can configure a SIP trunk with Teliax for dialing to the public telephone network. This topic describes the necessary configuration in Kerio Operator.
For more information, refer to Connecting to VoIP service providers (page 25).
Prerequisites
Teliax requires information about the IP address of your Kerio Operator instance in order to ensure a secure connection.
After setting up an account with Teliax, you should be given the following information from the provider, which is required in this configuration:
Server (IP or Domain such as test.ivy.teliax.com)
Login ID
Password
DID/Telephone Number
Channels (not needed for Operator configuration)
Configuration
1.
Log in to the web administration interface of Kerio Operator.
2.
Go to
Configuration > Call Routing
.
3.
Click
Add a SIP Interface
.
4.
Enter an interface name. For example,
Teliax
.
5.
Choose
New provider
and enter your telephone number (you may need to add a
1 at the beginning of the telephone number). Use a comma to separate multiple phone numbers.
6.
Click
Next
.
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7.
Choose the extension to receive incoming calls.
8.
Enter a dial out prefix if necessary.
9.
Click
Next
.
10.
Specify the hostname (Server) provided by Teliax ( test.ivy.teliax.com
) and the default port
5060
.
11.
Specify the
Username
(Login) and
Password
values as provided by Teliax.
12.
Enable
User ID differs from the telephone number
and enter your Teliax User ID (the same number that is in the
Username field).
13.
Enable the option
Register with registrar
.
14.
Click
Finish
.
15.
In order to transmit the correct CallerID information on outbound calls you must edit the interface that you’ve just created by doing the following:
Select the interface you just created (e.g. Teliax) and click
Edit
.
Click the
Advanced
tab.
Check the box
Use SIP user ID in REGISTER request only
.
Click
OK
to save.
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2.7.15 Connecting Kerio Operator to Vitelity
NOTE
This information is designed for Kerio Operator 2.3.5 and older.
You can configure a SIP trunk with
Vitelity for dialing to the public telephone network. This topic describes the necessary configuration in Kerio Operator. For more information refer to Connecting to VoIP service providers .
Prerequisites
After setting up an account with Vitelity, you should be given the following information from the provider:
Service
: Vitelity LLC VoIP
Username
:
*<your_user_name>
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Password
:
*<your_password>
Balance
:
$<account_balance>
The username and password above are for managing your Vitelity account. They are not used within Kerio Operator for any reason.
You then need to set up the following on the Vitelity admin console:
A
Sub Account
(your SIP account).
A DID Number.
The following screenshots serve as a guide:
Screenshot 3: Figure 1 – Vitelity DID Number Configuration
Screenshot 4: Figure 2 - Vitelity Sub Account (SIP) Configuration
After all the settings as described above, you need the following information for configuring Kerio Operator:
The telephone number (or numbers) assigned to you.
Login (sometimes referred to as Trunk #, User Name or User ID).
Password
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Register Server ( inbound34.vitelity.net
).
Outbound Server ( outbound.vitelity.net
).
Additional information needed for configuration under
Support > Generic Sip Support
(for example, a different
Register Server: sip34.vitelity.net)
.
Configuration
1.
Log in to the web administration interface of Kerio Operator.
2.
Go to
Configuration Call Routing
.
3.
Click
Add SIP Interface
.
4.
Enter an interface name (e.g. "
Vitelity
”).
5.
Choose
New Provider
and enter your telephone number (you may need to add a
1 at the beginning of the telephone number). Use a comma to separate multiple phone numbers.
6.
Click
Next
7.
Choose the extension to receive incoming calls
8.
Enter a dial out prefix if necessary.
9.
Click
Next
.
10.
Specify the hostname provided by Vitelity under
Support > Generic Sip Support
( sip34.vitelity.net
) and the default port 5060.
11.
Specify the
Username
(login and password values as you configured on Vitelity’s sub account settings (see Figure 2 above).
12.
Enable
User ID differs from the telephone number
and enter your Vitelity login (the same string that is in the username field).
13.
Enable the option to
Register with registrar
.
14.
Click
Finish
.
15.
In order to transmit the correct CallerID information on outbound calls, you must edit the interface that you created.
Do the following:
Select the interface you just created (e.g. Vitelity) and click
Edit
.
Click the Advanced tab
Check the box Use SIP user ID in REGISTER request only
Click OK to save
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2.7.16 How to configure Kerio Operator to connect to 802.cz
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
You need the following information for this configuration:
Your telephone number. You will have selected this number when you registered with 802.cz. You can also find the number if you log into your customer account at www.802.cz. The telephone number (in the national format, i.e. 9 digits) is also used as the authentication name.
Your SIP password. You can configure this in your customer account at
.
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Configuration
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
In the first screen, enter a description for the interface name (for example, '802.cz'), choose the option
New provider
and enter your telephone number in the national format (9 digits). For example, if your telephone number is in the international format
+ 420 333 123 456
, you would enter
333123456
.
4.
Click
Next
.
5.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out (for example, 0. The usual dial-out prefix in the Czech Republic).
6.
Click
Next
.
7.
Enter hlas.802.cz
as the
Domain (IP address/hostname)
. If you have a nomadic phone number, use sip.802.cz
instead.
8.
Enter your telephone number in the national format (9 digits, e.g.
333123456
) as the
Username
and enter your SIP password as the
Password
.
9.
Ensure
Required to register
is checked and click
Next
.
10.
Verify the information in the
Summary
section and click
Finish
.
Your
802.cz
connection is now configured. You can now test this, by placing some calls to your telephone number. This will verify that your SIP connection is working correctly.
2.7.17 How to configure Kerio Operator to connect to ActiveNetwork
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to configure connect
Kerio Operator
to Active Network using either a SIP truck also known as postpagato or, using account prepagato
.
Prerequisites
Before configuring
Kerio Operator
, you will require the following information:
Telephone number, as provided by Active Network in the
Area Riservata
on their website.
User-name of the SIP account.
Password
SIP Domain
The SIP account type postpagato or prepagato
.
If you have a firewall, make sure the SIP and RTP ports are open and properly routed to
Kerio Operator
.
Configuration
This process describes how to connect
Kerio Operator
to a SIP account with Active Network.
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1.
to the administration interface of your
Kerio Operator
and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter a description for the interface name. For example, activenetwork-prepagato
).
4.
Choose the option
New provider
and enter your telephone number.
With the option postpagato
For example, 507611878139.
, it is necessary have the number
5
in front of the phone number.
With the option prepagato
For example, 07611763399.
, the phone number doesn’t have any prefix in front of the number.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out.
7.
Click
Next
.
a.
Enter the Domain (IP address/hostname).
For an account prepagato use
VOIP.EUTELIA.IT
.
For an account postpagato use
SIP.TWT.IT
.
b.
Enter your
Active Network User Name
as the username and the
SIP-Password
as the password.
c.
Ensure
Required to register
is checked.
8.
Click
Next
to continue to the
Summary
screen.
9.
Verify the information and click
Finish
.
2.7.18 How to configure Kerio Operator to connect to Bandwidth.com
Learn how to configure Kerio Operator for use with a SIP trunk from Bandwidth.com.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
NOTE
Bandwidth.com requires that you run your VoIP PBX on a public IP address. Ensure that your Kerio Operator installation is secure.
Use strong passwords and configure the built-in firewall. Only your own phones and Bandwidth.com’s server should be allowed to communicate with your Kerio Operator PBX. With Kerio Operator 1.1, use the protection against SIP password guessing.
Prerequisites
Before starting this procedure, ensure you have:
Your account is configured for E.164 dialing (the international format with assume you use E.164 which is the default and also a slightly complex case.
+ at the beginning). In this topic, we
The telephone number assigned to you by Bandwidth.com. For example,
+1 234 555 0101
.
The IP address of Badwidth.com’s primary gateway. For example,
216.82.224.202
.
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Configuration
1.
to the administration interface of your Kerio Operator and go to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Key in the interface name. For example, bandwidth.com
.
4.
Choose the option
New provider
and key in your telephone number in the international format with
+1 at the beginning. For example,
+12345550101
.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls and key in the prefix that will be used to dial out. For example,
9
.
7.
Click
Next
.
8.
Key in the IP address (
216.82.224.202
) of Bandwidth’s primary gateway in the
Domain (IP address/hostname)
field.
9.
Uncheck
Require to register
, and leave the
username
and
password
fields empty. Bandwidth.com uses only IPbased authentication, so you must register your IP address with them.
10.
Click
Next
.
11.
Verify the information in the
Summary
section and click
Finish
to save the configuration of the interface.
You should now be able to receive incoming calls. But we need to modify the rules for outgoing calls so that outgoing calls work as well and dialed numbers are rewritten to the E.164 format.
We assume that if the PBX user dials a number with
91 at the beginning, they want to place a domestic call. If the number starts with
9011
, it’s an international call.
Dialing a
+ on a desktop phone is usually not easy, so we will add the
+ sign in the rewriting rules instead.
1.
Double-click on the rule for outgoing calls to edit it. In our example, the rule is displayed as
9... bandwidth.com
.
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2.
In the edit dialog, change the prefix to
91 and go to the
Called number
section.
3.
Set
Strip digits from left
to 1.
4.
In the
And add this prefix
field, key in +. Now, if someone dials
914084964500
, this number will be modified to
+14084964500
. This ensures we dial US-based numbers in the full E.164 format.
5.
Click
OK
to save the change.
6.
Click
Add…
on the
Call Routing
screen to add a second outgoing rule for prefix
9011
.
7.
Key in
9011 as the prefix and add the external interface for Bandwidth.com.
8.
Under
Called Numbers
, set
Strip digits from left
to
4 and set
And add this prefix
to
+
. On setting these fields, the prefix
9011 will be replaced with
+ and we will thus obtain a correct international E.164 format.
9.
Click
OK
to save the dialog.
Your Bandwidth.com trunk is now configured on Kerio Operator. Try running some test calls.
2.7.19 How to configure Kerio Operator to connect to Breezz (NL)
Learn how to configure
Kerio Operator
to connect to the Dutch provider Breezz.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
The telephone number assigned to you by
Breezz
. The number will include the Netherlands’ international country code (without
+ at the beginning). For example,
31718123456
. In the configuration, the phone number is also used as an authentication name.
The password for your SIP account.
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Configuration
1.
to the administration interface of
Kerio Operator
.
2.
Go to
Advanced Options
, tab
General
and set the
SIP User-Agent
string to
Kerio Operator
. Breezz will not let you register with the
Asterisk PBX
string.
3.
Go to the
Call Routing
screen.
4.
Click
Add a SIP Interface
.
5.
In the first screen, enter the interface name (for example,
Breezz-NL
), choose the option
New provider
and enter your telephone number (
31718123456 as per our example).
6.
Click
Next
.
7.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out. For example,
9
.
8.
Click
Next
.
9.
Enter sip.sipnl.net
to the Domain (IP address/hostname).
10.
Enter your telephone number (
31718123456
) as the username.
11.
Enter your SIP password.
12.
Ensure
Require to register
is checked.
13.
Click
Next
to go to the
Summary
screen.
14.
Verify the information and click
Finish
.
15.
Double-click the interface you have just created to edit the order of codecs.
16.
Move
G.711 A-law
and
G.711 U-law
to the top. It is recommended to this because
Breezz
now supports
G.726
and the translation from
G.711
to
G.726
reduces the audio quality noticeably. If your phones support
G.726
you may leave the codec order as it is and use
G.726
for the whole call path (
phone < > Operator < > Breezz
).
You are now ready to place and receive calls.
2.7.20 How to configure Kerio Operator to connect to DevopSys
Learn how to configure Kerio Operator to connect to Devopsys using a SIP trunk.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
Your telephone number.
The Devopsys SIP Domain or IP address.
Username and Password.
If you have a firewall, make sure the SIP and RTP ports are open and properly routed to Kerio Operator.
You must have communicated to Devopsys the IP address you are going to use to connect Operator to their services
– If it is behind firewall the IP address of the firewall will be used.
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Configuration
This process describes how to connect Kerio Operator to a SIP account with Devopsys.
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
In the first screen: a.
Enter a description for the interface name. For example,
Devopsys
.
b.
Choose the option
New provider
and enter your telephone number including the international dialing code. For example
0033 5 87 030300
.
4.
Click
Next
.
5.
Choose the extension to receive incoming calls, and enter the prefix that will be used to dial out, if you wish to use one.
6.
Click
Next
.
a.
Enter the SIP server in the
Domain (IP address/hostname)
.
b.
Enter your
Devopsys User Name
as the username and the
SIP-Password
as the password.
c.
Ensure
Required to register
is checked.
7.
Click
Next
to continue to the
Summary
screen.
8.
Verify the information and click
Finish
.
Your Devopsys connection is now configured.
2.7.21 How to configure Kerio Operator to connect to Exetel
Learn how to connect Kerio Operator to a SIP account with Exetel. Exetel can be reached at : http://www.exetel.com.au
.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
The telephone number assigned to you. The phone number will be used as an authentication name as well.
The password for your SIP account.
SIP proxy address ( example Exetel SIP server, sip1.exetel.com.au
.
Configuration
1.
to the administration interface of
Kerio Operator
.
2.
Go to the
Call Routing
screen.
3.
Click
Add a SIP Interface
.
4.
In the first screen, enter the interface name. For example,
Exetel
.
5.
Choose the option
New provider
and enter your telephone number.
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6.
Click
Next
.
7.
Choose the extension to receive incoming calls (queue, script, conference or group) and enter the prefix that will be used to dial out. For example,
9
.
8.
Click
Next
.
9.
Enter
Domain (IP address/hostname)
. For example, sip1.exetel.com.au
.
10.
Enter your telephone number as the
username
and your SIP password as
password
.
11.
Ensure
Required to register
is checked.
12.
Click
Next
.
13.
Verify the information in the
Summary
section and click
Finish
.
You are now ready to place and receive calls.
2.7.22 How to configure Kerio Operator to connect to fayn.cz
Learn how to connect Kerio Operator to a SIP account with Fayn. We assume that you already have a Fayn SIP account and know your SIP credentials.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
If you do not know the credentials, log in to your account at www.fayn.cz
, go to
Setup
and click
View
next to your SIP authentication name.
Before starting this procedure, ensure you have:
Your phone number as provided by fayn.cz.
Your password.
SIP proxy address. Currently, sip.fayn.cz
.
All of this information can be found at https://iz.fayn.cz/ after login, under the
Přehled MSN
menu and the
MSN
button.
Configuration
1.
to the Kerio Operator admin interface, go to the
Call Routing
section.
2.
Click
Add a SIP Interface
button.
3.
Name your new interface and enter your assigned phone number.
4.
Click
Next
.
5.
Select the desired internal extension (queue, script, conference or group) and optionally enter an outbound prefix.
6.
Click
Next
.
7.
Enter sip.fayn.cz
into the
Domain (IP address/hostname)
field.
8.
Enter
Username
(your assigned phone number) and your password.
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9.
Click
Next
.
10.
Verify the information in the
Summary
section and click
Finish
.
11.
Open the route configuration again, go to the
Codecs
tab and correct the supported codecs list to match those supported by Fayn (see your Fayn account for exact information: http://www.fayn.cz/pece-a-podpora/navody-anastaveni/
).
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12.
Press
OK
.
2.7.23 How to configure Kerio Operator to connect to ha-vel.cz
Learn how to connect Kerio Operator to a SIP account with ha-vel.cz.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
We assume that you already have a Ha-vel SIP account and know your SIP credentials. (If you do not know the credentials, login to your account at https://ha-loo.ha-vel.eu/cz/index.php
).
Before starting this procedure, ensure you have:
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Your phone number registered with ha-vel.cz. We assume it as
123456789 in our example below.
Your password.
SIP proxy address (currently ustredna.ha-vel.cz
).
These details can be found by logging into https://ha-loo.ha-vel.eu/cz/index.php
, under
Information / Informace
and
SIP/IAX Settings / Nastaveni SIP/IAX
sections .
Configuration
1.
to the Kerio Operator admin interface.
2.
Go to the
Call Routing
section and click
Add a SIP Interface
.
3.
Name your new interface and enter your assigned phone number.
4.
Click
Next
.
5.
Select the desired internal extension (queue, script, conference or group) and optionally specify the outbound prefix.
6.
Click
Next
.
7.
Enter ustredna.ha-vel.cz
into the
Domain (IP address/hostname)
field.
8.
Key in the assigned phone number into
Username
field and password into
Password
field.
9.
Click
Next
.
10.
Open route configuration again, go to the
Codecs
tab and remove unsupported codecs (SpeeX, G.722 and G.726
codecs).
11.
Click
OK
.
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2.7.24 How to configure Kerio Operator to connect to isphone
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
The telephone number assigned to you in E164 format. The phone number will be used as an authentication name as well.
The password for your SIP account.
SIP proxy address. For example, sip2.isphone.com.au
.
Configuration
1.
to the administration interface of
Kerio Operator
.
2.
Go to the
Call Routing
screen.
3.
Click
Add a SIP Interface
.
4.
Key in the interface name (for example, isphone
).
5.
Choose the option
New provider
and enter your telephone number.
6.
Click
Next
.
7.
Choose the extension to receive incoming calls (queue, script, conference or group) and enter the prefix that will be used to dial out. For example,
9
.
8.
Click
Next
.
9.
Set the
Domain (IP address/hostname)
, For example, sip2.isphone.com.au
.
10.
Enter your telephone number as the
username
and your SIP password as
password
.
11.
Ensure
Required to register
is checked.
12.
Click
Next
.
13.
Verify the information in the
Summary
section and click
Finish
.
You are now ready to place and receive calls.
2.7.25 How to configure Kerio Operator to connect to Megapath
NOTE
This information is designed for Kerio Operator 2.3.5 and older. For more information about creating and configuring a SIP interface in later versions, see
Connecting to VoIP service providers
.
Learn how to configure Kerio Operator with a Megapath SIP trunk.
Prerequisites
Before starting this procedure, ensure you have:
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Your telephone number(s).
The User-Name of the SIP account.
The password for the SIP account.
The host or IP of the SIP registration server (usually an EdgeMarc device).
Configuration
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter a description for the interface name. For example, megapath
.
4.
Choose the option
New provider
and enter your telephone number(s). For example,
4085555555
.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls and an optional outgoing prefix.
7.
Click
Next
.
8.
Enter the SIP registrar/proxy hostname (e.g.
192.168.1.2
).
9.
Keep the default port number,
5060
.
10.
Enter the username and password values as provided to you by Megapath.
11.
Ensure
Must register with the Registrar
is checked.
12.
Enable
User ID differs from the telephone number
, and specify the same value as your username
13.
Click
Finish
.
14. Edit your newly created SIP interface and from the
Advanced
menu enable
Use SIP user ID in REGISTER request only
option.
2.7.26 How to configure Kerio Operator to connect to MultiVoice
Learn how to configure Kerio Operator to connect to MultiVoice using a SIP trunk.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
Your telephone number.
The User name and password of the SIP account.
The SIP Domain.
If you have a firewall, make sure the SIP and RTP ports are open and properly routed to Kerio Operator.
Configuration
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
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3.
Enter a description for the interface name. For example,
MultiVoice
.
4.
Choose the option
New provider
and enter your telephone number including the international dialing code. For example,
0039 0171 699757
.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls, and optionally enter the prefix that will be used to dial out.
7.
Click
Next
.
8.
Enter the SIP Domain in
Domain (IP address/hostname
. For example, multivoice.multiwire.net
.
9.
Enter your
MultiVoice User Name as the
username
and the
SIP-Password as the
password
.
10.
Ensure
Required to register
is checked.
11.
Click
Next
.
12.
Verify the information in the
Summary
section and click
Finish
.
Your MultiVoice connection is now configured.
2.7.27 How to configure Kerio Operator to connect to netphone.cz
Learn how to connect Kerio Operator to a SIP account with Netphone CZ.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
We assume that you already have a Netphone SIP account and know your SIP credentials. If you do not know the credentials to login to your account on name.
www.netphone.cz
, go to Setup and click View next to your SIP authentication
Prerequisites
Before starting this procedure, ensure you have:
Your phone number, as provided by netphone.cz. For example,
123456789
.
Your password.
SIP proxy address. Currently, sip1.netphone.cz
.
These details can be found after logging in to https://admin.netphone.cz/prihlaseni
, under
Přehled tel. cisel
section and
MSN
button.
Configuration
1.
to the Kerio Operator admin interface, go to the
Call Routing
section and click
Add a SIP Interface
.
2.
Name your new interface and enter your assigned phone number on the 1st screen.
3.
Click
Next
.
4.
Select the desired internal extension (queue, script, conference or group) and optionally choose an outbound prefix.
5.
Click
Next
.
6.
Enter sip1.netphone.cz
in the
Domain (IP address/hostname)
field.
7.
Key in the
Username
(assigned phone number) and your
Password
.
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8.
Click
Next
.
9.
Verify the information in the
Summary
section and click
Finish
.
10.
Open route configuration and go to the
Codecs
tab and correct supported codecs list to match those supported by
Netphone. See your netphone.cz account, under the
Prehled tel. cisel
section, click
Upravit
button and enable
Expert mod
to get the list.
11.
Click
OK
.
2.7.28 How to configure Kerio Operator to connect to OrbTalk
NOTE
This information is designed for Kerio Operator 2.3.5 and older. For more information about creating and configuring a SIP interface in later versions, see
Connecting to VoIP service providers
.
Learn how to configure connect Kerio Operator to OrbTalk using a SIP truck.
Prerequisites
Before starting this procedure, ensure you have:
Your telephone number.
The user-name and password of the SIP account.
The SIP Domain
If you have a firewall, make sure the SIP (
UDP 5060 – 5090
) and RTP (
UDP 6000 – 65535
) ports are open and properly routed to Kerio Operator. Kerio Operator requires that ports
UDP 10000 - 19999 internally are open.
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Configuration
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter a description for the interface name. For example,
OrbTalk
.
4.
Choose the option
New provider
and enter your telephone number. For example,
0550935405
.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out.
7.
Click
Next
.
8.
Enter the SIP
registrar/proxy hostname
.
9.
Enter
5060 in the
Port Number
.
10.
Enter your OrbTalk User Name as the authentication name and the SIP-Password as the password. Typically the username will follow the format of
IPT-<country code and telephone number>
. For example,
IPT-
441223202130
.
11.
Ensure
Must register with the Registrar
is checked.
12.
Enable
User ID differs from the telephone number
, this will also be your User Name.
13.
Click
Finish
.
Your OrbTalk connection is now configured
2.7.29 How to configure Kerio Operator to connect to plusTEL in Denmark
Learn how to connect
Kerio Operator
to a SIP account with plusTEL.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
We assume that you already have a plusTEL SIP account and know your SIP credentials. (If you do not know the credentials, contact plusTEL.dk to receive this information.)
Before starting this procedure, ensure you have:
Your SIP authentication name.
Your SIP password.
The SIP registrar/proxy hostname.
Configuration
1.
to the administration interface of your
Kerio Operator
and go to the
Call Routing
screen. At least one extension is needed before creating the SIP Interface.
2.
Click
Add a SIP Interface
.
3.
Enter an interface name, choose the option
New provider
and enter your telephone number.
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4.
Click
Next
.
5.
Choose the extension to receive incoming calls on and if you like, enter the prefix that will be used to dial out.
6.
Click
Next
.
7.
Enter the
Domain (IP address/hostname)
.
8.
Enter your SIP
username
and
password
.
9.
Ensure
Required to register
is checked.
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10.
Click
Next
.
11.
Verify the information in the
Summary
section and click
Finish
.
Your connection to plusTEL is now configured.
plusTEL description plusTEL is a registered VoIP Service Provider in Denmark.
For further information contact:
SIP-Trunk
plusTEL ApS
TEL: +45 35294010 [email protected]
Kerio Distribution
MikroGraf as
TEL: +45 70222101 [email protected]
eStation ApS
TEL: +45 35294000 [email protected]
2.7.30 How to configure Kerio Operator to connect to sipgate.com
Learn how to connect Kerio Operator to a SIP account with sipgate.com.
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NOTE
This information is designed for Kerio Operator 2.3.5 and older. For more information about creating and configuring a SIP interface in later versions, see
Connecting to VoIP service providers
.
Prerequisites
Before starting this procedure, ensure you have:
Your telephone number, as provided by Sipgate. To find the number, log in to Sipgate administration interface at www.sipgate.com
and go to
Settings
.
The SIP-ID of your SIP account. You can locate this in the Sipgate administration interface, click
Settings > SIP Credential
, the SIP-ID will then be displayed. For example,
1234567a0
.
The SIP-ID (not the telephone number) is the most important identifier you require in order to connect over SIP. The
SIP-Password for your SIP account. This password is usually of the form
1ABCD2
.
Configuration
1.
to the administration interface of your Kerio Operator and navigate to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter a description for the interface name. For example, sipgate.com
.
4.
Choose the option
New provider
and enter your telephone number. With sipgate, it is not significant whether you include
1 at the beginning of the number or not.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls to and enter the prefix that will be used to dial out. For example,
9
.
7.
Click
Next
.
8.
Enter sipgate.com
as the SIP registrar/proxy hostname.
9.
Enter your sipgate SIP-ID as the authentication name and the SIP-Password as the password.
10.
Ensure
Must register with the Registrar or Proxy
is checked.
11.
Check
User ID differs from the telephone number
and enter your SIP-ID in the User ID field. This is the part where
Sipgate differs from most SIP providers who prefer to use the telephone number as the SIP user ID.
12.
Click
Finish
. Your sipgate connection is now configured.
You can now test the configuration by placing some calls to your telephone number. This will verify that your SIP connection is working correctly.
2.7.31 How to configure Kerio Operator to connect to Telephonic Canada
NOTE
This topic is for versions Kerio Operator 2.3.5 and older. For more information about creating and configuring a SIP interface in later versions, see
Connecting to VoIP service providers
.
Learn how to configure
Kerio Operator
with a SIP Trunk to Telephonic Canada
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NOTE
Watch the Configuring Kerio Operator with Telephonic Canada video.
Prerequisites
Before starting this procedure, ensure you have:
The telephone number or numbers assigned to you by Telephonic. Each number will include the Canadian’ international country code without
+ at the beginning. For example,
17785555555
.
If you have a firewall, make sure the SIP and RTP ports are open and properly routed to Operator.
A static IP, as the typical configuration will use IP based registration without requiring authentication.
You have provided Telephonic with your static IP address.
Configuration
1.
to the administration interface of
Kerio Operator
.
2.
Go to the
Call Routing
screen.
3.
Click
Add a SIP Interface
.
4.
Enter the interface name. For example,
Telephonic
.
5.
Choose the option
New provider
and enter your telephone number. For example,
17785555555
. In case of multiple numbers, use comma separation as noted in the dialog.
6.
Click
Next
.
7.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out. For example,
9
.
8.
Click
Next
.
9.
In the SIP Registrar dialog, specify the hostname provided by Telephonic ( kerio.telephonic.ca
) and the default port
5060
. Leave the Username and password fields empty, and disable both checkboxes for registration and user ID.
NOTE
Telephonic Canada provides a free testing account (with DID) and $5.00 credit for customers evaluating Kerio
Operator.
2.7.32 How to configure Kerio Operator to connect to Voicepulse.com
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to connect Kerio Operator to a SIP account with Voicepulse Inc. We assume that you already have a Voicepulse
SIP account and know your SIP credentials. (If you do not know the credentials, log in to your account at www.voicepulse.com
, go to
Setup
and click
View
next to your SIP authentication name.
Prerequisites
Before starting this procedure, ensure you have:
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Your telephone number assigned to you by Voicepulse. For example
+1 555 123 4567
.
Your SIP authentication name – note this is not the user name you use to log into Voicepulse web interface. The SIP authentication name is typically something like
VaSBFLM12a
".
Your SIP password. Again, this is not the password you use to login to Voicepulse web interface. It is the second part of your SIP credentials. The password is typically something like u2TYRUa878
.
Configuration
1.
to the administration interface of your Kerio Operator and go to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter a interface name, choose the option
New provider
and enter your telephone number with
1 at the beginning.
4.
Click
Next
.
5.
Choose the extension to receive incoming calls on and enter the prefix that will be used to dial out. For example,
9
.
6.
Click
Next
.
7.
Enter jfk-primary.voicepulse.com
as the
Domain (IP address/hostname)
. Use sjc-primary.voicepulse.com
if you are closer to San Jose, CA than to New York.
8.
Enter your SIP authentication name and password.
9.
Ensure
Required to register
is checked.
10.
Click
Next
.
11.
Verify the information in the
Summary
section and click
Finish
.
Your connection to Voicepulse is now configured.
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2.7.33 How to configure Kerio Operator to connect to VOIP-Unlimited
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to connect Kerio Operator to a SIP account with VOIP Unlimited. We assume that you already have a VOIP unlimited SIP account and know your SIP credentials.
Prerequisites
Before starting this procedure, ensure you have:
Your phone number and username, as provided by VOIP Unlimited.
Your password.
SIP proxy address. Currently, sip.voip-unlimited.net
.
All of this information is emailed to you from VOIP-Unlimited.
Configuration
1.
to the Kerio Operator admin interface, go to the
Call Routing
section and click
Add a SIP Interface
.
2.
Name your new interface and enter your assigned phone number.
3.
Click
Next
.
4.
Select the desired internal extension (queue, script, conference or group) and optionally enter an outbound prefix.
5.
Click
Next
.
6.
Enter sip.voip-unlimited.net
in the
Domain (IP address/hostname)
field, enter your username (not your phone number) into the
Username
field and your password into the
Password
field.
7.
Click
Next
.
8.
Verify the information in the
Summary
section and click
Finish
.
9.
Open the route configuration and go to the
Codecs
tab and correct the supported codecs list to match those supported by VOIP-Unlimited. The supported codecs are G729, G711a and G711u.
10.
Click
OK
.
2.7.34 How to configure Kerio Operator to connect to VoipVoice
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to configure connect
Kerio Operator
to VoipVoice using a SIP truck.
Prerequisites
Before starting this procedure, ensure you have:
Your telephone number.
The user name and password of the SIP account.
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The SIP Domain.
If you have a firewall, make sure the SIP and RTP ports are open and properly routed to
Kerio Operator
.
Configuration
1.
to the administration interface of your Kerio Operator and navigate to the Call Routing screen.
2.
Click
Add a SIP Interface
.
3.
Enter a description for the interface name. For example,
VoipVoice
.
4.
Choose the option
New provider
and enter your telephone number. For example,
0550935405
.
5.
Click
Next
.
6.
Choose the extension to receive incoming calls and enter the prefix that will be used to dial out.
7.
Click
Next
.
8.
Enter the SIP domain in the
Domain (IP address/hostname)
field.
9.
Enter your
VoipVoice User Name as the
username
and the
SIP-Password as the
password
.
10.
Ensure
Required to register
is checked.
11.
Click
Next
.
12.
Verify the information in the
Summary
section and click
Finish
.
Your VoipVoice connection is now configured.
2.7.35 How to configure Kerio Operator to connect to Xphone.cz
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to connect Kerio Operator to a SIP account with Xphone CZ. We assume that you already have a Xphone SIP account and know your SIP credentials. (If you do not know the credentials, login to your account at www.xphone.cz
, go to
Setup
and click
View
next to your SIP authentication name.
Prerequisites
Before starting this procedure, ensure you have:
Your phone number, as provided by xphone.cz. For exzample,
123456789
.
Your username and password.
SIP proxy address. Currently, sip2.xphone.cz
.
All of this information can be found on logging in to http://www.xphone.cz
, under
Nastaveni / Settings
and then
Prihlasovaci udaje / login
credentials.
Configuration
1.
to the Kerio Operator admin interface, go to
Call Routing
section and click
Add a SIP Interface
.
2.
Name your new interface and enter your assigned phone number.
3.
Click
Next
.
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4.
Select the desired internal extension (queue, script, conference or group) and optionally enter an outbound prefix.
5.
Click
Next
.
6.
Enter sip2.xphone.cz
in
Domain (IP address/hostname)
, enter your username (not your phone number) in
Username
and your password in
Password
.
7.
Click
Next
.
8.
Verify the information in the
Summary
section and click
Finish
.
9.
Open the route configuration and go to the
Codecs
tab and correct the supported codecs list to match those supported by Xphone. See your xphone.cz account for exact information https://www.xphone.cz/zone_user_settings_ codecs.php
).
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10.
Press
OK
.
2.7.36 How to connect Kerio Operator to Skype Connect
NOTE
This functionality was last redesigned for Kerio Operator 2.4 and newer.
Prerequisites
Before starting this procedure, ensure you have:
The SIP User ID for your Skype Connect profile. To find this ID, log to your Skype Manager account, view the details of your Skype Connect profile and navigate to Authentication Details. The SIP User is usually a 14-digit number that starts with
9905
.
The password generated by Skype for your Skype Connect profile. You will find it in the same Skype Manager page as the SIP User ID.
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The phone number you have associated with Skype Connect profile. Skype lets you choose a number in about 20 countries. For example, a phone number in USA would look like
+1 408 555 0123
.
Configuration
1.
to the administration interface of your Kerio Operator and go to the
Call Routing
screen.
2.
Click
Add a SIP Interface
.
3.
Enter an interface name, choose the
New provider
option and enter your telephone number. For example,
14085550123
.
4.
Click
Next
.
5.
Choose the extension to receive incoming calls and enter a prefix that will be used for external calls. For example,
9
.
6.
Click
Next
.
7.
In the
Domain (IP address/hostname)
field, enter sip.skype.com
.
8.
Enter your SIP User ID in the
Username
field and enter the
Password
.
9.
Ensure
Required to register
is checked.
10.
Click
Next
.
11.
Verify the information in the
Summary
section and click
Finish
.
You should be able to receive calls at your telephone number after finishing these steps. You should also be able to call numbers within the country in which you have chosen your phone number.
However, you have to do a few additional steps to be able to call international numbers.
Skype requires that you send international numbers in the E.164 format with
+ in front of the country code. In our example, the dial-out prefix is
9 and we will add the prefix
900 for international calls. The approach is as follows:
1.
Go to the
Call Routing
screen again and click
Add
in
Routing of outgoing calls
.
2.
In the
Add Outgoing Route
dialog, enter
900 as the route prefix and add your Skype Connect interface as created before.
3.
Modify the values for the
Called number
so that 3 digits are stripped off as we want to remove the prefix
900
) and replace is with
+
.
4.
Click
OK
to save the route.
Now, as an example, if you want to call a number in the Netherlands, you would dial
90031… and this would get translated this to
+31… and will send the call to your Skype Connect interface.
The Skype™ name, Skype Connect™ and their logos are trademarks of Skype.
2.8 Gateways
This section helps you connect via various gateways.
2.8.1 Configuring Kerio Operator and Cisco SPA8800 for calls over an analog telephone line
2.8.2 Configuring Kerio Operator and Grandstream GXW 4104/4108 for calls over analog telephone lines
2.8.4 Configuring Kerio Operator and Grandstream HT503 for calls over analog lines
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2.8.6 Configuring Kerio Operator and Well/Yeastar NeoGate TB400 for calls between SIP and EuroISDN
2.8.7 Configuring Kerio Operator and Well/Yeastar NeoGate TG200 for calls between SIP and GSM
2.8.8 Configuring Kerio Operator and Yeastar NeoGate TE100 for calls over analog lines
2.8.9 Configuring PRI telephone service through the Digium VoIP Media Gateway
2.8.10 Connection with Linksys SPA3102 analog (FXS/FXO) to SIP gateway
2.8.1 Configuring Kerio Operator and Cisco SPA8800 for calls over an analog telephone line
Cisco SPA8800 is an analog-to-SIP gateway equipped with 4 FXS and 4 FXO telephone ports.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to configure the SPA8800 with Kerio Operator to place and receive phone calls over the FXO interface (analog telephone line).
Prerequisites
We assume that your Kerio Operator is up and running and you have at least one internal extension.
We also expect:
Your SPA8800 is already connected to your LAN and you have access to its web administration. To discover the device’s IP address or change the basic network settings, you can connect an analog phone to the
Phone 1
port on the
SPA and use the device’s simple IVR system as described in the SPA8800 Quick Start Guide.
Your PSTN line is connected to the
Line 1
port of the SPA device.
Additionally, we have describe the configuration of a single line. If you are about to use several lines on the SPA, you need to repeat the configuration steps up to 4 times. The SPA8800 uses a separate SIP/UDP port for every analog line, so you need one SIP interface in Kerio Operator for each line in use. The default port numbers are
5061 for Line 1,
5161 for Line 2, 5261 for Line 3, and
5361 for Line 4.
Configuring Kerio Operator
NOTE
The PSTN phone number
5550199
, Kerio Operator IP address
10.1.2.95
and SPA8800 IP address 10.1.2.200
are all sample values and are used as examples to explain the configuration process.
1.
In the administration interface, go to
Call Routing
.
2.
Create a new SIP interface with
5550199 as the external number.
3.
Click
Next
.
4.
In the second screen, choose an extension to receive calls from this interface and set a dial-out prefix (for example 9).
5.
In the third screen, set the following:
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a.
Set the IP address of the SPA device. For example,
10.1.2.200
b.
Set the username. Kerio Operator uses to authenticate with the SPA. For example, spa8800
.
c.
Set a password, for example pass5550199
.
d.
Clear the
Required to register
option. The SPA is not able to behave as a SIP registrar.
For more information, refer to Connecting Kerio Operator to TelePacific (page 52).
6.
In the last screen, select
Edit details of the created interface
and click
Finish
.
After you finish the configuration of the SIP interface, the
Edit External Interface
dialog box opens:
1.
Go to the
SIP Details
tab.
2.
In the
Outbound Proxy
field, set the IP address of the SPA device with a port number 5061. For example,
10.1.2.200:5061
.
3.
Leave the
Inbound Proxy
field empty.
4.
Click
OK
to save your changes.
Configuring SPA8800
1.
Connect to the web interface of your SPA device. For example,
10.1.2.200
.
2.
Go to the administration section (log in as the administrator, if needed) and click
Voice
and then
Advanced
— in our example, we reach the URL http://10.1.2.200/admin/voice/advanced
.
3.
Go to the
Line 1
configuration page. All subsequent steps (except the step 11) will be done on this page.
4.
Verify that
Line Enable
is set to
yes
.
5.
Under SIP Settings, verify that
SIP Port
is set to
5061
. Set the
Auth INVITE
field to
yes
.
6.
Scroll down to the
Proxy and Registration
section:
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a. Proxy contains Kerio Operator’s IP address. For example,
10.1.2.95
.
b.
Set
Use Outbound Proxy
to
no
.
c.
Set
Register
to
no
.
d.
Set
Use OB Proxy In Dialog
to
no
.
e.
Set
Make Call Without Reg
to
yes
.
f.
Set
Ans Call Without Reg
to
yes
.
5.
Scroll to the
Subscriber Information
section: a.
Set
Display Name
to some non-empty string. Use something like External call or Analog device to make it easier for your users to distinguish calls coming through the SPA device.
b.
Set
User ID
to your external telephone number, for example
5550199
.
c.
Set
Password
to your chosen authentication password, for example pass5550199
.
d.
Set
Use Auth ID
to
yes
.
e.
Set
Auth ID
to your chosen authentication user name. For example, spa8800
.
Screenshot 5: Configuring Proxy and Subscriber Information in SPA8800
6.
Scroll to the
Dial Plans
section. Set
Dial Plan 8
to
S0<:[email protected]>
. Here,
5550199 is the external phone number and
10.1.2.95
is the IP address of Kerio Operator.
7.
Now move to the VoIP-To-PSTN Gateway Setup section. Set the
VoIP Caller Default DP
field to
none
.
8.
Scroll to the
PSTN-To-VoIP Gateway Setup
section: a. PSTN CID Number Prefix should be empty.
b. PSTN Caller Default DP must be
8
. (DP stands for Dial Plan — see step 6 above).
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NOTE
Note about Caller ID: On a typical PSTN line, the Caller ID (aka CLIP or CID) is usually transmitted using the FSK modulation between the first and the second ring of the incoming call. If your PSTN line provides the Caller ID service, set
PSTN CID For VoIP CID
to
yes
but also increase the value in
PSTN Answer Delay
(under
FXO Timer
Values
) so that the SPA device can hear the 2 rings before starting the VoIP call to Kerio Operator.
9.
Go to the
FXO Timer Values (sec)
section: a.
Set
VoIP Answer Delay
to
0
.
b.
Set
PSTN Answer Delay
to
0
. If you want to transfer the PSTN Caller ID to the VoIP side, enter a number (in seconds) that approximately corresponds to 2 rings on your PSTN line.
10.
Save the configuration by clicking on
Submit All Changes
at the bottom of the configuration page. Note that the device might need 20-30 seconds to save and apply the new configuration.
11.
If you enabled the Caller ID detection in step 8, go to the
Regional > Miscellaneous
and check that the fields
Caller
ID Method
and
Caller ID FSKStandard
correspond with the standard used by the local telco company. The most usual method is
Bellcore
—
bell202
, but you may need to ask the telco about the Caller ID encoding method they are using.
Both Operator and the SPA8800 are now configured and you can place some test calls. If you are not satisfied with the volume of the sound transmitted from the analog side to VoIP or vice versa, you can return to the
Line 1
configuration page, scroll to
International Control
and change the values for
SPA To PSTN Gain
and/or
PSTN To SPA Gain
.
2.8.2 Configuring Kerio Operator and Grandstream GXW 4104/4108 for calls over analog telephone lines
Grandstream GXW 4104 is an analog-to-SIP gateway with four FXO ports. The GXW 4108 model has eight FXO ports.
Prerequisites
To complete the configuration, you need:
Kerio Operator up and running.
At least one internal extension.
The Grandstream gateway connected to your LAN.
One analog phone connected to the first port of the GXW gateway.
In the example below:
The Kerio Operator IP address is 10.1.2.95.
The Grandstream gateway's IP address is 10.1.2.200.
The phone number of connected phone is 1234567.
The internal extension's number is 100.
NOTE
The example below uses GXW 4104, firmware version 1.4.1.5. The GXW gateway runs without a SIP password in a peer-to-peer configuration. Use this configuration in a safe local network.
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Kerio Operator configuration
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Type a name for the interface (for example, the provider's name). The name must not contain spaces or special characters and must be unique.
4.
Select
New provider
.
5.
In the
With external number
field, type the number
1234567 and click
Next
.
6.
Select an extension that receives all calls, For example, (
100
.
7.
Optionally, in the
Prefix to dial out
field, type a prefix for outgoing calls and click
Next
.
8.
In the
Domain (IP address/hostname)
field, type the IP address of the Grandstream GXW gateway,
10.1.2.200
.
9.
Deselect the
Required to register
option and click
Next
.
10.
Verify the information in the
Summary
section and click
Finish
.
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Grandstream GXW configuration
To configure the GXW 4104 model, use the ch1-4 parameter. For a GXW 4108 model configuration, use the parameter ch1-8
.
1.
In the Grandstream GXW administration interface, go to
Accounts > Account 1 > General Settings
.
2.
In the
SIP Server
field, type the IP address of (
10.1.2.95
).
3.
In the
Account Active
field, select
Yes
.
4.
Optionally, set up your
Account Name
and
Outbound Proxy
.
5.
Save your settings.
The web interface now asks whether you want to reboot to apply the new configuration. You can reboot now or continue with the configuration and reboot later.
1.
Go to
Accounts > User Account
and verify that the configuration form is empty.
2.
Go to
Settings > Channel Settings > Calling to VoIP
.
a.
Set
User ID
to ch1-4:1234567;
. Calls that come to any of the four analog ports are send to Kerio Operator as if the number called was 1234567. To differentiate among the analog ports, type additional external numbers (for example,
1234568
) in the SIP interface in Kerio Operator and set
User ID
to ch1-
2:1234567;ch3-4:1234568
. This configuration maps analog ports 1 and 2 to the number
1234567 and ports 3 and 4 to the number
1234568
.
b.
In the
SIP Server
field, set ch1-4:pl;
.
c.
In the
SIP Destination Port
field, set ch1-4:5060;
.
3.
Go to
FXO Lines > Settings > Port Caller ID Setting
.
a.
In the
Number of Rings Before Pickup
field, set ch1-4:2;
. Analog lines usually transmit the caller ID between the first and second ring. If you want to detect the caller ID, you must allow at least two rings before
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accepting the call.
b.
In the
Caller ID Scheme
field, set the option to the method used by your analog line provider. For example, for Bellcore use ch1-4:1;
.
c.
In the
Caller ID Transport Type
field, set ch1-4:1; to transmit the caller ID in the SIP
From
field.
4.
Go to
FXO Lines > Dialing > Dialing to PSTN
, and in the
Stage Method (1/2)
field, set ch1-4:1;
.
5.
Reboot the gateway.
After the reboot, make several test calls to verify the configuration.
NOTE
To change the volume of the sound transmitted between the analog line and the SIP, go to
FXO Lines > Settings >
Port Voice Settings
and modify the audio gain values.
2.8.3 Configuring Kerio Operator and Grandstream GXW4224 to use analog phones for internal extensions
Grandstream GXW4224 is an analog-to-SIP gateway that allows you to connect up to 24 analog phones. Similar models with 16, 32, and 48 FXS ports exist.
Prerequisites
Before going into actual process, we assume the following:
Kerio Operator is up and running.
You have at least one other SIP phone that allows you to place a test call.
Grandstream gateway is connected to your LAN and you have access to its web configuration interface.
At least one analog phone is connected to the Grandstream gateway (to the 1st FXS port);
Kerio Operator configuration
1.
In the administration interface, go to
Extensions
.
2.
Create the extensions you are about to assign to the analog phones. In this example, we create just a single extension
1101
.
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Screenshot 6: Edit Extension dialog
NOTE
The SIP username can be the same as extension number but we use the SIP username
1101p1 to show that the username can be different. Also, write down the SIP password or copy it to the clipboard.
Grandstream GXW4224 Configuration
1.
Connect to the Grandstream’s web interface.
2.
Go to
Profile 1
(see screenshot below). The Grandstream gateway lets you configure up to 4 SIP servers.
3.
Configure Kerio Operator as the 1st server (Profile 1).
4.
In the Profile 1, check that
Profile Active
is set to
Yes
.
5.
Key in Kerio Operator’s IP address as the
Primary SIP Server
. For example,
10.1.2.95
.
6.
Check that the field
DNS Mode
is set to the value
A Record
.
7.
Check that the
Local SIP port
is set to
5060
.
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Screenshot 7: Profile 1 screen
8.
Scroll down to the bottom of the
Profile 1
page and click
Update
.
9.
Go to the screen
FXS Ports
.
10.
Key in the SIP identifiers, password, and server profile for each of the 24 analog ports (or 16, 32, 48 ports, respectively). To configure the extension 1101, we do the following for
FXS Port
1: a.
Enter
1101p1 into both
SIP User ID
and
Authenticate ID
fields.
b.
Enter the extension’s password.
c.
Enter
1101 into the
Name
field.
d.
Select
Profile 1
.
Screenshot 8: FXS Ports
11.
Scroll down to the bottom of the screen and click
Update
.
12.
Reboot the gateway.
Testing
When the reboot is complete, you should see the extension registered in Kerio Operator’s
Extensions
grid. Also the
Grandstream’s
Status
screen should show the registered extensions. You can now place some test calls between analog
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phones connected to the gateway and your other SIP phones.
2.8.4 Configuring Kerio Operator and Grandstream HT503 for calls over analog lines
The
Grandstream HT503 is an analog-to-SIP ATA device with one FXO and one FXS port. You can use this device with
Kerio Operator to make calls over the telephone network.
Prerequisites
To complete the configuration, you need:
Kerio Operator up and running.
At least one internal extension.
Telephone line connected to the FXO port.
The Grandstream HT503 connected to the same LAN as Kerio Operator.
The example in this topic uses the following inputs:
The Kerio Operator IP address is
10.1.2.95
.
The Grandstream HT503's IP address is
10.1.2.200
.
The external number from the provider is
123456
.
The SIP password used for the FXO configuration on the Grandstream HT503 is pass1234
.
Connecting to a network with another DHCP server
The Grandstream HT503 runs a DHCP server on the LAN port. If you have another DHCP server in your network, connect the device to your network via the WAN port and enable web access to the administration interface on that port.
To enable the access:
1.
After the device boots, connect an analog phone to the FXS port.
2.
Press
*** to access the voice menu.
3.
Press
12 and then
9
.
To hear the IP address of the device, press
*** and then
02
.
Configuring the Grandstream HT503
To configure the Grandstream device, start by configuring the FXO port:
1.
In the administration interface of the device, go to the
FXO PORT
section.
2.
In the
Primary SIP Server
field, type the Kerio Operator IP address. For example,
10.1.2.95
.
3.
In the
SIP User ID
and
Authenticate ID
fields, type the external number. For example,
123456
.
4.
In the
SIP Registration
field, select
No
.
5.
Optionally, in the
Caller ID Scheme
field, select the method used by your provider to detect the caller's ID for incoming calls.
6.
In the
Caller ID Transport Type
field, select
Relay via SIP From
.
7.
In the
Number of Rings
field:
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To detect the caller's ID for incoming calls, type
2
.
If your provider does not offer Caller ID, type
1
.
8.
In the
PSTN Ring Thru FXS
field, select
No
.
9.
In the
Wait for Dial-Tone
field, select
No
.
10.
In the
Stage Method
field, type
1
.
11.
Click
Update
and then
Apply
to save your changes.
After you finish configuring the FXO port, configure the rest of the settings:
1.
Go to the
BASIC SETTINGS
section.
2.
Go to the
Unconditional Call Forward to VOIP
section.
3.
In the
User ID
field, type the external number (
123456 in the example).
4.
In the
SIP Server
field, type the Kerio Operator IP address (
10.1.2.95
in the example).
5.
In the
SIP Destination Port
field, type
5060
.
6.
Click
Update
and then
Apply
to save the changes.
Configuring Kerio Operator
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Type a name for the interface (for example, the name of the gateway). The name must not contain spaces or special characters and must be unique.
4.
Select
New provider
.
5.
In the
With external number
field, type the external number (
123456
) and click
Next
.
6.
Select an extension that receives all calls.
7.
Optionally, in the
Prefix to dial out
field, type a prefix for outgoing calls and click
Next
.
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8.
In the
Domain (IP address/hostname)
field, type the IP address of the Grandstream device (
10.1.2.200
).
9.
In the
Username
field, type the external number.
10.
In the
Password
field, type the SIP password used for the FXO configuration on the Grandstream HT503. For example, pass1234
.
11.
Clear the
Required to register
option and click
Next
.
12.
Select the
Edit details of the created interface
option and click
Finish
.
After Kerio Operator finishes the configuration wizard, the Edit External Interface (SIP) dialog box opens:
1.
Go to the
SIP Details
tab.
2.
In the
Outbound proxy
field, type
10.1.2.200:5062
.
3.
In the
Inbound proxy
field, type
10.1.2.200:5062
.
4.
Click
OK
to save your changes.
After you complete the configuration, make some test calls to verify the connection between Kerio Operator and
Grandstream HT503.
2.8.5 Configuring Kerio Operator and WellTech 2504/WellGate 2504 to use analog phones for internal extensions
WellTech 2504 is an analog-to-SIP gateway with 2 Ethernet ports and 4 FXS ports that allow you to connect up to 4 analog phones. It is sold as WellGate 2504 in some markets.
Learn how to integrate Kerio Operator with WellTech 2504/WellGate 2504.
Prerequisites
Before starting the configuration, we assume:
Your Kerio Operator is up and running. In the example below the Kerio Operator’s IP address has been used as
10.1.2.95
.
You have at least one other SIP phone that allows you to place a test call.
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Your WellTech 2504 is connected to your LAN and you have access to its web configuration interface.
At least one analog phone is connected to WellTech 2504 (to port Tel 1).
NOTE
The WellTech gateway runs a DHCP server on its LAN interface by default. If you already have a DHCP server running in your LAN, do not connect the gateway’s LAN interface to your network straight away. Following the device’s quick installation guide, you can first connect a single computer to the WellTech’s LAN interface and set up networking as needed for your LAN.
Kerio Operator configuration
1.
In the administration interface, go to
Extensions
.
2.
Create the extensions you are about to assign to the analog phones. In this example, we create just a single extension
1001.
Screenshot 9: Edit Extension dialog
NOTE
Because of the limitations in how the WellTech gateway handles SIP identifiers, the SIP username must be the same as the extension number (
1001
). Also. take a note of SIP password or copy it to the clipboard.
WellTech 2504 Configuration
1.
Connect to the WellTech’s web interface.
2.
Go to
FXS Settings > SIP Proxy
.
3.
In the
Primary Proxy Server
, type Kerio Operator’s IP address.
4.
Check that the
Primary Proxy Server Port
is set to
5060
.
5.
Click
Apply
.
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Screenshot 10: SIP Proxy screen
6.
Go to
FXS Settings > FXS Line
and click the edit icon for the first line. The edit icons are in the first column of the table. The line edit screen should appear.
7.
Scroll to the bottom of the screen, set the field
Register Type
to
Register
and type the extension number (
1001
) into the
TEL No
field.
8.
Type the extension’s SIP username into the field
User ID
(it must be the same as the extension number 1001).
9.
Type (or paste) the extension’s SIP password.
10.
Click
Apply
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Screenshot 11: FXS Setting
If you are using multiple lines, configure them in a similar way. The unused lines should be set as inactive (the field
Line
State
).
The WellTech gateway needs to be restarted to start using the new configuration. Go to
Maintenance
and reboot the device.
Testing
The WellTech gateway may need about a minute to reboot. As soon as it is up and running, the extensions should appear as registered in Kerio Operator’s
Extensions
grid. You can now place some test calls between the analog phones and another SIP phone.
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2.8.6 Configuring Kerio Operator and Well/Yeastar NeoGate TB400 for calls between SIP and EuroISDN
NOTE
This information is designed for Kerio Operator 2.3.5 and older.
For more information about creating and configuring a SIP interface in newer versions, see
Connecting to VoIP service providers
.
Yeastar NeoGate TB400 (sold under the Well brand in some countries) is a SIP-to-ISDN gateway. The gateway can be equipped with 2 or 4 BRI ports and hence it supports up to 8 parallel calls.
Prerequisites
Before starting the configuration, we assume:
Kerio Operator is up and running.
You have at least one other SIP phone that allows you to place a test call.
Your NeoGate TB400 is connected to your LAN and you have access to its web configuration interface.
You have one EuroISDN line connected to the 1st BRI port on the NeoGate.
You have two external phone numbers assigned to your EuroISDN line.
You know the signaling type for your ISDN line (point-to-point or point-to-multipoint).
In the example below:
The Kerio Operator IP address is
10.1.2.95
.
The NeoGate TB400 IP address is 10.1.2.200.
The two external phone numbers are
300123456 and
300123457
.
NeoGate TB400 configuration
Connect to the web administration interface of your TB400 and do the following steps:
1.
Go to the
BRI Settings > Module List
screen and open the edit dialog for module BRI1.
2.
Set
Signaling
to
BRI-CPE or
BRI-CPE-PTMP depending on whether your line uses point-to-point or point to multi-point signaling, respectively.
3.
Switch to the
DOD Settings
section in the edit dialog and key in the first external phone number (
300123456
) in both
the DOD start from
and
Associated number start from
fields.
4.
Click
Add DOD
and add your second external number (
300123457
) as done in the previous step.
5.
Click
Save
to close the edit dialog.
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Screenshot 12: Configuring external numbers
6.
Go to the
SIP Settings > Trunk
screen.
7.
Select
VoIP Account
.
8.
Set the
Type
field to
SIP and
Transport
to
UDP
.
9.
Type your first external phone number (
300123456
) into the
Account
field.
10.
Type some complex password into the
Password
field.
11.
Click
Save
.
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Screenshot 13: Configuring VoIP account in TB400
12.
Click
Apply Changes
.
IMPORTANT
You may encounter a situation with gateway not using the DOD numbers on the BRI interface correctly until restarted. So you may also want to restart NeoGate at this point
Kerio Operator configuration
1.
Connect to your Kerio Operator administration GUI and go to the
Call Routing
screen.
2.
Create a new SIP interface. Enter your external numbers separated with comma in the first screen of the
Add SIP Interface
wizard. For example,
300123456,300123457
.
3.
In the second screen of the
Add SIP Interface
wizard, select an extension to receive calls from this interface and set a dial-out prefix (in our example, let’s use 9 as the prefix).
4.
Enter the following values in the third screen: a.
Key in the gateway’s IP address (
10.1.2.200
) in the
Hostname or IP address
field.
b. Port number should be
5060
.
c.
The
Username
should be your phone number (
300123456
).
d.
Key in your TB400 VoIP account password.
e. Must register with the Registrar or Proxy should remain checked.
You can now place some test calls. Use the prefix you have assigned to the new SIP interface to call via NeoGate TB400.
2.8.7 Configuring Kerio Operator and Well/Yeastar NeoGate TG200 for calls between SIP and GSM
Yeastar NeoGate TG200 (Well NeoGate TG200 in some countries) is a SIP-to-GSM gateway. The gateway can be equipped with one or two GSM modules and supports at most two parallel calls.
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NOTE
This information is designed for Kerio Operator 2.4 and newer.
Prerequisites
To complete the configuration, you need:
Kerio Operator up and running.
At least one other SIP phone that allows you to place a test call.
The NeoGate TG200 gateway connected to your LAN.
At least one SIM card inserted into NeoGate TG200 and the SIM card’s PIN number in the NeoGate’s web configuration.
In the example below:
The Kerio Operator IP address is 10.1.2.95.
The NeoGate gateway IP address is 10.1.2.200.
The SIM card number is 700123456.
Configuring NeoGate TG200
Connect to the web administration interface of your NeoGate TG200:
1.
Go to
SIP Settings > Trunk
.
2.
Select the
VoIP Account
option.
3.
Set the
Type
field to
SIP
and
Transport
to
UDP
.
4.
In the
Account
field, key in the SIM card number.
5.
In the
Password
field, key in a password. Use this password when
6.
Click
Save
.
Screenshot 14: Configuring VoIP account in TG200
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7.
Go to
Route Settings > Outgoing Routes
.
8.
Add a new outgoing route.
NOTE
The default route in NeoGate TG200 does not work when calling numbers in the international format with “+” at the beginning.
If you need the international format when calling back, add another route as a workaround:
1.
Go to
Route Settings > Outgoing Routes
.
2.
Add a new outgoing route.
3.
In the
New Outgoing Route
dialog box, key in a new route name in the
Route Name
field.
4.
In the
Dial Pattern
field, key in the
+X.
string.
5.
Move at least one GSM module from
Available Trunks
to
Selected
.
6.
Click
Save
.
Screenshot 15: The route to handle the international number format in TG200
Configuring Kerio Operator
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
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3.
Key in a name for the interface. The name must not contain spaces or special characters and must be unique.
4.
Select
New provider
.
5.
In the
With external number
field, key in the SIM card number and click
Next
.
6.
Select an extension that receives all calls.
7.
In the
Prefix to dial out
field, key in a prefix for outgoing calls (
7 in our example) and click
Next
.
8.
In the
Domain (IP address/hostname)
field, key in the IP address of the NeoGate TG200 gateway.
9.
In the
Username
field, key in the SIM card number and key in the
configured for the VoIP account in the gateway
10.
Select the
Required to register
option and click
Next
.
11.
Verify the information in the
Summary
section and click
Finish
.
You can now place some test calls. To make a call via the NeoGate TG200 gateway, use the prefix for outgoing calls configured for the gateway.
NOTE
Kerio Operator adds the interface prefix and extends the number (7+4411234567 in our example). To call back, you can dial the extended number because of the additionally configured outgoing route.
2.8.8 Configuring Kerio Operator and Yeastar NeoGate TE100 for calls over analog lines
Yeastar NeoGate TE100 is an analog-to-SIP gateway with a single E1/T1/J1 port that supports the PRI standard. You can use this device with Kerio Operator to make calls over the telephone network.
Learn how to integrate Kerio Operator with Yeastar NeoGate TE100.
Prerequisites
To complete the configuration, you need:
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Kerio Operator up and running.
At least one internal extension.
The Yeastar NeoGate gateway connected to the same LAN as Kerio Operator.
In the example below:
The Kerio Operator IP address is
192.168.62.107
.
The Yeastar NeoGate gateway's IP address is
10.1.2.200
.
The trunk of numbers from the provider is
555 12xx
.
The internal extension number is
100
.
Configuring Yeastar NeoGate TE100
1.
In the administration interface of the Yeastar NeoGate gateway, click
Gateway
.
2.
In the
Digital Trunk
section, click to open the trunk configuration.
3.
In the
Mode Type
field, select:
E1 in Europe.
T1 in the USA.
3.
In the
Linecoding
field, select:
HDB3 in Europe.
B8ZS in the USA.
4.
In the
Framing
field, select:
Enable CRC4 in Europe.
ESF in the USA.
5.
In the
Switch Type
field, select:
Euro ISDN in Europe.
national in the USA.
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6.
Click
Save
.
After you finish configuring the trunk, you need to verify that the gateway is connected:
1.
Click the
Status
button in the upper right corner.
2.
Go to
System Status > E1/T1 Status
.
3.
Verify that
Alarm
indicates
Connect
.
After verifying the gateway connection, continue with the configuration:
1.
Go to
VoIP Settings > VoIP Trunk
and remove all default VoIP trunks.
2.
Click
Add VoIP Trunk
to open the
Add New Account
dialog box.
3.
In the
Trunk Type
field, select
Service Provider
.
4.
Key in a name for the VoIP trunk. For example,
My Operator
.
5.
Key in the IP address (
192.168.62.107
) of Kerio Operator.
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6.
Click
Save
.
7.
Go to
Route Settings > Route List
.
8.
Click to edit the
E1_to_SIP route.
9.
In the
Send Call Through
field, select
ServiceProvider — MyOperator and click
Save
.
10.
Click to edit the
SIP_to_E1 route.
11.
In the
Call Comes in From
field, select
ServiceProvider — MyOperator and click
Save
.
12.
Click
Apply Changes
in the upper right corner.
Configuring Kerio Operator
1.
In the administration interface of Kerio Operator, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface (for example, the name of the gateway). The name must not contain spaces or special characters and must be unique.
4.
Select
New provider
.
5.
In the
With external number
field, key in the number
555 12xx and click
Next
.
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6.
Select an extension (
100
) that receives all calls.
7.
Optionally, in the
Prefix to dial out
field, key in a prefix for outgoing calls and click
Next
.
8.
In the
Domain (IP address/hostname)
field, key in the IP address (
10.1.2.200
) of the Yeastar NeoGate gateway.
9.
Verify the information in the
Summary
section and click
Finish
.
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2.8.9 Configuring PRI telephone service through the Digium VoIP Media Gateway
You can connect Kerio Operator to a Primary Rate Interface (PRI) telephone service. This requires a gateway device or adapter to convert the voice media into a signal that Kerio Operator can process. The following topic covers the necessary steps to configure Kerio Operator with the
Digium G200 VoIP Media Gateway
.
Installing the Digium VoIP Media Gateway on the network
The device obtains an IP address automatically when connecting to the network. You can identify the device's IP address from your network DHCP server. If you use Kerio Operator as the DHCP server you can locate the leased IP address from
Configuration > Network > [DHCP leases button].
Connecting to the Digium VoIP Media Gateway
1.
Input the IP address of the device into the web browser of a management computer located on the same network.
You must connect using a secure (HTTPS) type of connection.
2.
Login to the device as
Admin
with the password
Admin
3.
After logging into the device, you can assign a static IP address and configure a new administration password. Refer to the Gateway User’s Manual for details.
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Configuring the Digium VoIP Media Gateway
Adding a SIP endpoint
A SIP endpoint defines the credentials that Kerio Operator uses to register with the gateway device.
1.
Locate
Configuration > SIP endpoints
2.
Add a new SIP endpoint and input the following parameters:
Name
: Custom name for the SIP endpoint (e.g.,
operator
).
Username
: The SIP username that Kerio Operator uses to register.
Password
: The SIP password that Kerio Operator uses to register.
Registration
: Select
Endpoint registers with this gateway
.
All other parameters should use the default setting.
Adding call routing rules
The Digium device can support multiple SIP endpoints and PRI/BRI interfaces. Therefore it is necessary to create rules that define where to route incoming and outgoing calls. Locate
Configuration > Call Routing Rules
.
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Create a rule to direct calls from the PRI/BRI interface to the SIP endpoint:
From
: Choose the PRI/BRI port (e.g.,
port1
).
To
: Choose the SIP endpoint (e.g.,
operator
).
Match
: Choose
All
.
DID
Manipulation:
None
.
Create another rule to direct calls from the PRI/BRI interface to the SIP endpoint. Use the same parameters as the previous rule, with the
From
and
To
values in reverse order.
NOTE
If you have multiple PRI/BRI interfaces you should create a Call Routing Group that includes each connected PRI interface. Refer to this group when configuring Call Routing Rules.
Registering Kerio Operator to the Digium device
Locate
Configuration > Call Routing
in the Kerio Operator administration. Add a SIP interface and configure the following parameters:
Interface name
: A label for the SIP interface.
New provider with external number
: Input the phone number(s) assigned by your telephone service provider.
Incoming calls
: Specify where to route incoming calls.
Outgoing calls
: Define a prefix if desired.
Domain, hostname or IP address
: IP address of the Digium device.
Username
: The username assigned to the SIP endpoint.
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Password
: The password assigned to the SIP endpoint.
Required to register with Registrar
: Enable this option.
After you add the SIP interface, edit the interface and go to the
SIP details
tab. Locate the
Authentication
username field and define the username of the SIP endpoint.
NOTE
You may need to contact your telephone service provider to ensure that they do not remove digits from the telephone number when routing incoming calls to your PRI service. You can refer to the security log to verify the details of rejected calls.
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Checking connectivity status
You can review the status of the SIP endpoints and PRI ports from the administration of the Digium device. Locate the
Diagnostics
dialog. Review the
Network, SIP Endpoints
, and
T1/E1
Interfaces to verify successful connectivity of all configured items.
2.8.10 Connection with Linksys SPA3102 analog (FXS/FXO) to SIP gateway
Linksys SPA3102 is an analog-to-SIP gateway equipped with one FXS and one FXO telephone port.
NOTE
This information is designed for Kerio Operator 2.4 and newer.
Learn how to configure SPA3102 gateway with Kerio Operator over the FXO interface.
Prerequisites
To complete the configuration, you need:
Kerio Operator up and running.
At least one internal extension.
The PSTN line connected to the
Line
port of the SPA3102 device.
The SPA3102 gateway connected to your LAN through only one port.
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IMPORTANT
The Linksys SPA3102 gateway has its own DHCP server in the
Ethernet
(LAN) interface and a DHCP client in the
Internet
(WAN) interface. If you already have another DHCP server in your LAN, do not connect the Ethernet interface of the gateway. Follow the steps in the installation guide of the device.
After the installation of the device:
1.
Connect the
Internet
port to your LAN.
2.
Connect an analog phone to the
Phone
port of the SPA3102 gateway.
3.
Dial
**** to access the voice menu.
4.
Dial
110# to obtain the IP address of the device.
5.
To enable web administration on the WAN port of the SPA3102 gateway, dial
7932# 1#
.
In the example below:
The Kerio Operator IP address is
10.1.2.95
.
The Linksys SPA3102 gateway IP address is
10.1.2.200
.
The PSTN number is
123456
.
Configuring Kerio Operator
1.
In the Kerio Operator administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP Interface
.
3.
Key in a name for the interface. The name must not contain spaces or special characters and must be unique.
4.
Select
New provider
.
5.
In the
With external number
field, key in the PSTN number and click
Next
.
6.
Select an extension that receives all calls.
7.
In the
Prefix to dial out
field, key in a prefix for outgoing calls (
9
) and click
Next
.
8.
In the
Domain (IP address/hostname)
field, key in the IP address of the Linksys SPA3102 gateway.
9.
Key in an username ( spa3102
) and a password ( pass123456
).
10.
Disable the
Required to register option
and click
Next
. The gateway does not behave as a SIP registrar, so Kerio
Operator must authenticate before each call.
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11.
Select the
Edit details of the created interface
option and click
Finish
.
After you finish the configuration, the
Edit External Interface
dialog box opens:
1.
Go to the
SIP Details
tab.
2.
In the
Outbound proxy
field, key in the IP address of the gateway and the port,
10.1.2.200:5061
. The
Inbound proxy
field leave empty.
3.
Click
OK
to save your settings.
Configuring the Linksys SPA3102 gateway
1.
Connect to the Linksys SPA3102 web administration interface and go to
Voice > Advanced
:
2.
Go to
Line 1
, in the
Line Enable
field select
no
, and click
Submit All Changes
to save your settings.
3.
Go to
PSTN Line
and in the
Line Enable
field select
yes
.
4.
In the
SIP Port
field, key in
5061
.
5.
Go to the
Proxy and Registration
section: a.
In the
Proxy
field, key in the Kerio Operator IP address.
b.
In the
Use Outbound Proxy
field, select
no
.
c.
In the
Register
field, select
no
.
d.
In the
Use OB Proxy In Dialog
field, select
no
.
e.
In the
Make Call Without Reg
field, select
yes
.
f.
In the
Ans Call Without Reg
field, select
yes
.
5.
Go to the
Subscriber Information
section:
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a.
In the
Display Name
field, key in a new name (
External Call in our example).
b.
In the
User ID
field, key in the PSTN number.
c.
In the
Password
field, key in the password
configured for the SIP interface in Kerio Operator
.
d.
In the
Use Auth ID
field, select
yes
.
e.
In the
Auth ID
field, key in the username
configured for the SIP interface in Kerio Operator
.
6.
Go to the
Dial Plans
section and in the
Dial Plan 8
field key in the
S0<:[email protected]> string.
7.
Go to the
VoIP-To-PSTN Gateway Setup
section: a.
In the
VoIP-To-PSTN Gateway Enable
field, select
yes
.
b.
In the
VoIP Caller Auth Method
field, select
HTTP Digest
.
c.
In the
One Stage Dialing
field, select
yes
.
d.
In the
VoIP Caller Default DP
field, select
none
.
8.
Go to the
VoIP Users and Passwords (HTTP Authentication)
section: a.
In the
VoIP User 1 Auth ID
field, key in the username
configured for the SIP interface in Kerio Operator
.
b.
In the
VoIP User 1 Password
field, key in the password
configured for the SIP interface in Kerio Operator
c.
In the
VoIP User 1 DP
field, select
none
.
9.
Go to the
PSTN-To-VoIP Gateway Setup
section: a.
In the
PSTN-To-VoIP Gateway Enable
field, select
yes
.
b.
In the
PSTN Caller Auth Method
field, select
none
.
c.
In the
PSTN Ring Thru Line 1
field, select
no
.
d.
Leave the
PSTN CID Number Prefix
option blank.
e.
In the
PSTN Caller Default DP
field, select
8
.
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NOTE
If your PSTN line provides the Caller ID service, in the
PSTN CID For VoIP CID
field, select
yes
, and increase the value for
PSTN Answer Delay
(see step below), so the SPA3102 gateway can transmit the Caller ID using the FSK modulation between the first and the second ring of the call before the start of the VoIP call to Kerio Operator.
10.
Go to the
FXO Timer Values (sec)
section: a.
In the
VoIP Answer Delay
field, key in
0
.
b.
In the
PSTN Answer Delay
field, key in
0
.
NOTE
To transfer the PSTN Caller ID to the VoIP side, key in the number of seconds that represents two rings on your PSTN line.
11.
Click
Submit All Changes
to save your settings.
2.9 Kerio Operator API
The Kerio Operator API enables you to programmatically access your Kerio Operator server to integrate with third-party solutions or write scripts to automate specific tasks. The API provides all actions available in the client and administration interfaces of the product. For example, you can add/remove users, update IP address groups, read logs, manage time ranges, and much more.
Licensing
The API is part of Kerio Operator and is governed by the product’s license. In summary, the license to use Kerio Operator also entitles you to use the API on your server. Third-party software vendors may distribute their applications both for free or for profit. There is no royalty fee for using the API.
Get involved
A dedicated forum is available for sharing your ideas, questions, answers, and other feedback with the developer community. You can participate in the forum at forums.kerio.com
.
Introduction
The API is accessible via a secure, session based HTTP connection. All communication is formatted as a JSON string and directed to a designated URL. For example, a request for the Kerio Operator administration API may go to: https://server.example.com:4021/admin/api/jsonrpc/
The URL to access the API consists of the following parts:
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URL part
https:// server.example.com
:4021
/admin/api/jsonrpc/
Title
Protocol
Hostname
Port
File path
Description
The API uses HTTPS
The Internet accessible hostname (domain name) of your server
The port number for accessing the API
The file path of the JSON interface
The Kerio Operator API is accessible via two separate interfaces that provide different functionality. The Kerio Phone interface provides phone functionality such as adding favorites, reviewing recent calls, or setting up call forwarding. The
Kerio Operator Administration interface configures all server based functionality such as adding users, configuring SIP interfaces, or assigning extensions. The port number and file path vary depending on the interface.
Refer to the table below:
Product interface
Kerio Phone
Kerio Operator Administration
Port
443
4021
File path
/myphone/api/jsonrpc/
/admin/api/jsonrpc/
Prerequisites
To access the API, you need a Kerio Operator user account. Depending on the interface, this may be a standard user account, or an administrator account with elevated permissions. The type of account you use to authenticate determines your permissions when accessing the API. In order to communicate with the API you need a framework for processing the requests and managing the network connection. A common framework for these actions is PHP and cURL.
Authentication
The API uses a two step authentication process. The first authentication step involves a username and password, the same type of account used to access the Kerio Operator interface. After successful authentication, the API returns a randomized token, and a session cookie. All subsequent requests to the API must include the token and session cookie in the HTTP headers of the connection. The method for handling the session cookie and token depends on the framework used in your application. Refer to the example below for information on how you can manage authentication to the API using cURL.
API Requests
All requests to the API must be encoded as a JSON string. The JSON string is an array consisting of the JSON version, an identifier, the API method, and the parameters of the query. You can use a web browser to inspect the JSON request and response for any interaction with the product interface. Refer to
Inspecting the Kerio Operator API communication in a web browser for details.
Example API Communication
The following example uses PHP and cURL to get the next available extension number.
//Custom variables
$api_cookie="/tmp/kerio-api-cookie";
$api_url="https://operator.example.com:4021/admin/api/jsonrpc/";
$api_user="admin-api";
$api_password="secretpassword”;
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//Initialize the cURL request
$ch = curl_init(); curl_setopt($ch, CURLOPT_URL, $api_url); curl_setopt($ch, CURLOPT_HTTPHEADER, "Content-Type:application/json"); curl_setopt($ch, CURLOPT_RETURNTRANSFER, true); curl_setopt($ch, CURLOPT_SSL_VERIFYHOST, false); curl_setopt($ch, CURLOPT_SSL_VERIFYPEER, false); curl_setopt($ch, CURLOPT_COOKIEJAR, $api_cookie); curl_setopt($ch, CURLOPT_COOKIEFILE, $api_cookie);
//Server login request to obtain the API token and session cookie
$api_login = array(
'jsonrpc' => '2.0',
'id' => 1,
'method' => 'Session.login',
'params' => array(
'userName' => $api_user,
'password' => $api_password,
'application' => array(
'name' => 'Sample app',
'vendor' => 'Kerio',
'version' => '1.0'
)
);
) curl_setopt($ch, CURLOPT_POSTFIELDS, json_encode($api_login));
$return=json_decode(curl_exec($ch),true);
$token=$return['result']['token'];
//Verify the token, otherwise return the JSON response and exit the script.
if ($token=="") {printf(htmlspecialchars($json_return)); exit();}
//Add the token to the cURL headers curl_setopt($ch, CURLOPT_HTTPHEADER, array("Content-Type:application/json","X-Token:".$token));
//Get the next available extension number
$api_query = array(
'jsonrpc' => '2.0',
'id' => '1',
'method' => 'Extensions.getNextValidNumber',
'params' => array()
); curl_setopt($ch, CURLOPT_POSTFIELDS,json_encode($api_query));
$return=json_decode(curl_exec($ch),true);
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$next_ext=$return['result']['extensionNumber']; curl_close($ch);
2.9.1 Inspecting Kerio Operator API communication in a web browser
You can use a web browser to inspect the API communication used in Kerio Operator. This is helpful in case you want to learn how to adapt functionality into your custom application.
The web interfaces of Kerio Operator use XMLHttpRequest (XHR) to exchange JSON formatted data. Browsers display this type of activity in the network details section of the developer view. You can find each JSON formatted request and the corresponding response of any action you perform in the interface.
1.
to the Kerio Phone or Kerio Operator administration interface.
2.
Enable the developer view in your browser.
3.
Perform the action you want to build into your application.
4.
Locate the JSON requests and responses in your browser's developer view.
Using Google Chrome to obtain the API request for adding a user in Kerio Operator
1.
In Google Chrome, enable the
Developer Tools
.
2.
Log in to the Kerio Operator administration and create a new user.
For more information, refer to Creating user accounts (page 192).
3.
In the Google Chrome developer view, go to the
Network
panel. See
Network panel overview for details.
4.
Locate the resources named jsonrpc/ in the
Requests Table
and select the first item. Note that you can also filter the table results to display only API resources. In the filter box, type the string "jsonrpc".
5.
In the preview area of the resource, select the
Headers
tab.
6.
Expand
Request Payload
to view the formatted JSON request.
7.
Browse each JSON resource and examine the payload details until you locate the resource that corresponds to your action.
8.
Click
view source
to show the unformatted JSON string.
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Screenshot 16: JSON resources showing in Developers Tools
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3 Using
This section contains information about:
3.1 Hardware phones and devices
3.3 CRM integration and desktop dialers
3.1 Hardware phones and devices
This section helps you configure and use hardware phones.
3.1.1 Hardware telephone basic usage
3.1.2 Configuring BLF on Polycom phones
3.1.3 Configuring Cisco / Linksys SPA phones to support more than three callers in a conference
3.1.4 Configuring Snom M300/M700 with Kerio Operator
3.1.5 Configuring the Aastra 6755i IP Phone with Kerio Operator
3.1.6 How to configure Busy Lamp Field (BLF) on Cisco SPA500S
3.1.7 How to configure Busy Lamp Field (BLF) on snom phones
3.1.8 How to configure Busy Lamp Field (BLF) on Well phones
3.1.9 Linksys/Cisco SPA: Setting the TFTP address without using the DHCP parameter 66
3.1.1 Hardware telephone basic usage
Learn how to use common functions of hardware phones supported by Kerio Technologies ).
Using loud speaker
Every telephone has a special button for loud speaker (speaker button). You can usually press this button either before dialing or anytime during a call.
Using Do not Disturb
If you select Do not Disturb option (DnD), the phone will generate the busy tone to inform the other party that you are not available.
Learn how to configure DnD in various supported hardware phones.
Cisco IP Phone 7960/7940
1.
Press the
Settings
button
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.
2.
Scroll to
Call
Preferences and press
Select
.
3.
Scroll to
Do Not Disturb
.
4.
Press
Yes
.
Cisco SPA
1.
Press
more
.
2.
Press
DnD
.
Polycom IP 33x
1.
Press
Menu
.
2.
Select
Features
.
3.
Select
Do not disturb
.
For unblocking incoming calls, use the same sequence.
Forwarding calls
Most hardware phones support call forwarding. However, you can also set it in
Kerio Phone too.
Conference calls
Most hardware phones also support conference calls. However, it is much easy and simple to use
.
Transferring calls
When you want to transfer a call, you usually have two options:
Attended transfer -you can connect with the third party, find out if the person is on the phone and make an announcement.
Blind transfer - the call is transferred without any announcement.
Learn how to configure these options in various supported hardware phones.
Cisco IP Phone 7960/7940
Attended transfer
1.
Initiate or answer the call.
2.
Press
more
.
3.
Press
Transfer
.
4.
Dial the number to which you want to transfer the call.
5.
When the dialed number rings, press
Transfer
again, or wait for the other party's answer, announce the call and then press
Transfer
.
6.
Hang up if the party accepts the call.
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If the party refuses the call, return to the original call by pressing the
Resume
softkey.
Blind transfer
1.
Initiate or answer the call.
2.
Press
more
.
3.
Press
BlndXfr
.
4.
Dial the number to which you want to transfer the call.
5.
Terminate the call.
Cisco SPA
Attended transfer
1.
Initiate or answer the call.
2.
Press
Trnsfer
.
3.
Dial the number to which you want to transfer the call.
4.
Wait for the connection.
5.
Press
Trnsfer
again.
6.
Hang up.
Blind transfer
1.
Initiate or answer the call.
2.
Press
Trnsfer
.
3.
Dial the number to which you want to transfer the call.
4.
Press
Trnsfer
again.
5.
Hang up.
Polycom IP 33x
Attended transfer
1.
Initiate or answer the call.
2.
Press
Trans
.
3.
Dial the number to which you want to transfer the call.
4.
Wait for the connection.
5.
Press
Trans
again.
6.
Hang up.
Blind transfer
1.
Initiate or answer the call.
2.
Press
Trans
.
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3.
Press
Blind
.
4.
Dial the number to which you want to transfer the call.
5.
Hang up.
Snom 300
Attended transfer
1.
Initiate or answer the call.
2.
Press
L1/L2
key (hold).
3.
Dial the number to which you want to transfer the call.
4.
Wait for the connection.
5.
Press
Transfer
key.
6.
Press
OK
key.
7.
Hang up.
Blind transfer
1.
Initiate or answer the call.
2.
Press
Transfer
key.
3.
Dial the number to which you want to transfer the call.
4.
Press
OK
key.
5.
Hang up.
Snom 320/821
Attended transfer
1.
Initiate or answer the call.
2.
Press
Hold
key.
3.
Dial the number to which you want to transfer the call.
4.
Wait for the connection.
5.
Press
Transfer
key.
6.
Press
OK
key.
7.
Hang up.
Blind transfer
1.
Initiate or answer the call.
2.
Press
Transfer
key.
3.
Dial the number to which you want to transfer the call.
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4.
Press
OK
key.
5.
Hang up.
3.1.2 Configuring BLF on Polycom phones
BLF (Busy Lamp Field) is a light placed on a telephone device. BLF notifies you if another extension is busy or not.
Setting BLF on Polycom telephones is not as easy as on other telephones. You need to prepare a configuration file and upload it to Kerio Operator.
Preparing a configuration file
Prepare a file with the following content:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?><attendant attendant.resourceList.1.address="sip:[email protected]" attendant.resourceList.1.label="Ten" attendant.resourceList.2.address="sip:[email protected]" attendant.resourceList.2.label="Eleven"/> <call call.directedCallPickupMethod="legacy" call.directedCallPickupString="*8"/>
This example file enables two BLF buttons. Button one monitors extension
10 on Kerio Operator with IP address
192.168.12.91
. This button is labeled
Ten
. A second button monitors extension
11 on same Kerio Operator and has label
Eleven
.
is set to
*8
.
Upload the file to
/var/tftp/polycom-0004f223510b-manual.cfg
where
0004f223510b is a phone hardware (MAC) address.
Uploading the configuration file to Kerio Operator
NOTE
The file must be uploaded via SSH using SCP.
Enabling SSH in Kerio Operator
Follow these instructions:
1.
In the administration interface, go to
Status > System Health
.
2.
Click
Tasks
while pressing the
Shift
key.
3.
Select
Enable SSH
.
4.
Connect to Kerio Operator via SCP (use for example WinSCP for Windows). For access, use username root and password of a Kerio Operator administrator.
5.
Upload the file.
3.1.3 Configuring Cisco / Linksys SPA phones to support more than three callers in a conference
The Cisco / Linksys SPA phone models support 3-way calling, which allows up to three callers in a conference. In some cases, you may require more than three callers. In this scenario, administrators can configure
. To further simplify the workflow, you can adjust the conference button of Cisco / Linksys phones to utilize a pre-configured conference line. This allows the operator of the Cisco / Linksys phone to move callers into the conference, without requiring the caller to hang up and call back into the conference extension.
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Configuring dedicated conference extensions
Each person who may need to incorporate this type of conferencing should be assigned their own
. For example, Alice may have extension 220, and Bob has extension 221. Their dedicated conference
extensions may be 3220 (for Alice) and 3221 (for Bob).
Adjusting the phone's conference button
Access the phone's web administration by entering its IP address into a browser (from a computer on the same network).
You can obtain the phone's IP address through the soft menu of the device, or from the 'Extensions' dialog in the Kerio
Operator administration. After entering the phone's administration, follow these steps:
1.
Click the
Admin login
link in the top right corner (If the phone is automatically provisioned, you may be prompted to login).
2.
If you are required to login, the user is 'admin' and the password is the master password assigned in the C
onfiguration > Provisioned Phones
dialog (Additional information is available in
Configuring automatic phone provisioning
).
3.
Click the
advanced
link.
4.
Go to the
Ext 1
tab.
5.
Locate the
Conference Bridge URL
input field.
The value of the
Conference Bridge URL
is the extension of the conference at (@) the host / IP of Kerio Operator. For example, Alice's dedicated conference is
3220
, so her conference bridge URL would be [email protected] (where example.com is the hostname of your Kerio Operator). After inputting the value, click
Submit All Changes
at the bottom of the page and allow the phone to reboot.
Operating conferences from the phone
By specifying a conference bridge URL, the action of the conference soft key will change, however the functional behavior remains the same. To place callers into the conference, follow these steps:
1.
Press the
conf
soft key while on an active call (this will place the caller on hold)
2.
Make a new outgoing call
3.
Once connected to the new party, press the 'confLx' soft key. This will move all parties to the conference.
4.
Repeat steps 1 - 3 for each additional caller.
3.1.4 Configuring Snom M300/M700 with Kerio Operator
Snom M300/M700 is a SIP-to-DECT base station working with M-series wireless handsets.
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NOTE
This information is designed for Kerio Operator 2.4 and newer.
Kerio Operator provisions Snom M300 and M700 base stations, but you must register handsets and assign internal extensions to them to make and receive calls.
Prerequisites
The base station connected to Kerio Operator.
At least one Snom handset.
At least one internal extension assigned to the phone.
To assign an extension to the phone:
1.
Go to
Provisioned Phones
.
2.
Select a phone and click
Edit
.
3.
Click
Add
and assign an extension.
4.
Click
OK
to save your changes.
For more information, refer to How to add a phone (page 171).
Configuring the base station and the handset
After Kerio Operator provisions the base station and you receive the IP address and password, you need to register the handsets and then assign the internal extensions in the base station's administration interface.
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Registering the handset
To see the handset in the base station's administration interface, you must register the handset first:
1.
Open the main menu of the Snom handset and go to
Connectivity
.
2.
Select
Register
.
3.
Key in a password of the handset and click
OK
.
The handset then registers to the base station.
Assigning the internal extension to the handset
After you register the handset to the base station, you need to assign an extension to the registered handset:
1.
Log in to the administration interface of the Snom base station. See the official Snom wiki for more details.
2.
Key in admin as the username and a password.
3.
Click
OK
.
4.
In the administration interface, go to
Extensions
.
5.
In the
Extensions
table, click the number of the extension you want to assign.
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6.
In the Select
Handset(s)
table, select the handset you want to assign that extension to.
7.
Click
Save
.
The display name and the number of the internal extension display on the main screen of the handset. Make a test call to verify the configuration of the handset.
Configuring time zones
To display your local time on handsets, you must configure the time zone in the administration interface of the base station:
1.
Log in to the administration interface of the Snom base station.
2.
Go to
Timezone
.
3.
Disable the
Set timezone by country/region
option.
4.
Set the
Timezone
.
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5.
Click
Save
.
Your handsets now automatically display the local time.
3.1.5 Configuring the Aastra 6755i IP Phone with Kerio Operator
This topic covers the basic configuration and usage of an Aastra 6755i IP phone with Kerio Operator.
NOTE
As of Kerio Operator version 2.3, this phone model can be automatically provisioned. Refer to the system requirements for a complete list of auto-provisioned phones.
Provisioning a new line (SIP account)
To configure a new line, log into the web administration of the phone by typing the phone's IP address into your browser. The default login for Aastra phones is admin/22222
.
Navigate to
Advanced Settings > Line 1
and input the following values:
Screen Name - The name that will appear on the phone's LCD.
Phone Number - Your extension.
Authentication Name - Your SIP login name (e.g. 230p1).
Password - Your SIP password.
NOTE
Before configuring the line, make sure you've created an extension in Kerio Operator.
Proxy Server - The IP Address or hostname of Kerio Operator.
Registrar Server - The IP Address or hostname of Kerio Operator.
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Configuring Busy Lamp Field (BLF)
Busy Lamp Field (BLF) allows the phone operator to view the status of other extensions, using an LED indicator. There is also an associated button next to the indicator, which acts as a speed dial to the extension being monitored.
The Aastra 6755i includes 12 programmable buttons / indicators that can be used for various functions, including BLF.
The actions for these buttons are located under
Softkeys and XML
, and
Programmable Keys
. Both configuration dialogs offer the same functionality, however the softkeys label and accessibility can change based on the phone's status, while the labels of the programmable keys are static.
To set a button for BLF, select BLF from the pull down menu under the type column. The value will be the extension you want to monitor. In case of call parking, this will be any one of the assigned parking slots (e.g. *51).
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Parking a call
Call parking is a feature that allows you to transfer a call to a temporary holding extension, which can then be joined from another phone. Before anyone can park a call, a range of parking slots must be defined in Kerio Operator. For configuration details refer to the KB topic
Configuring and using call parking
. To park an active call, press the Xfer
(transfer) soft key. Then press the button which has been designated as a parking slot. Wait until you've been connected to the parking slot, then press again the Xfer button. The caller is then put into the parking slot, and will hear hold music.
You can then hang up the phone.
NOTE
If nobody picks up the call within the defined timeout period (40 seconds by default), the caller will be bounced back to the person who parked the call.
Joining a parked call
If you have configured BLF with a parking slot, the LED will indicate that there is a parked call on that parking slot. To join the call, simply press the button and the phone will speed dial to the extension of that parking slot.
3.1.6 How to configure Busy Lamp Field (BLF) on Cisco SPA500S
Busy Lamp Field (BLF) feature allows users to monitor several other extensions. Sometimes, the term "Direct Station
Selection” is used for the same functionality. State of the monitored extensions is usually indicated by a series of LED lights with buttons.
1.
Idle state – LED is green
2.
Error state – LED is orange
3.
Ringing state – LED is red (blinking)
4.
Busy or Connected state – LED is red
The "Call Pickup” function is usually configured together with BLF allowing, for example, a receptionist to pickup ringing calls with a press of a single button.
Kerio Operator configuration
Busy Lamp Field
There is no configuration needed to make BLF work.
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Call Pickup
Enable the Call Pickup feature and configure an extension (e.g. "**”). When dialing "**10” while extension 10 is ringing, the call will be redirected to your phone.
Phone configuration
This guide has been tested on a Cisco IP Phone SPA508G (firmware 7.4.8a) with an attendant console SPA500S (hw 1.0.6, sw 2.0.2).
1.
Open the phone administration in your browser (eg. http://192.168.1.10).
2.
Login as administrator and open advanced configuration.
3.
Select the
Attendant Console
screen.
4.
Set
Subscribe Expires
to
1800
.
5.
Set
Unit 1 Enable
to yes
.
6.
Set
Unit 2 Enable
to yes if you have a second SPA500S unit.
7.
Set
Server Type
to
Asterisk
.
8.
Set
Attendant Console Call Pickup Code
to the extension of Directed Call Pickup followed by a
# sign. When pressing a button, the # sing will be replaced by the extension number. Use "**#” for Directed Call Pickup at "**”.
9.
Configure the unit keys. Use fnc=blf+sd+cp;sub=10@$PROXY to monitor extension 10.
10.
Don't forget to save the configuration.
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Call Parking
To park a call, use the xfer (transfer) soft key. Then dial to a designated parking slot (e.g. *53). If you have configured BLF keys to monitor parking slots, you can simply press the key which is monitoring the parking slot (e.g. *53). After you hear the announcement you will then hear the hold music. You can then press the xferLx (blind transfer) soft key to join the caller into the parking slot. The caller is then parked, and you can hang up the call.
3.1.7 How to configure Busy Lamp Field (BLF) on snom phones
Busy Lamp field (BLF) feature allows users to monitor several other extensions. Sometimes the term
Direct Station
Selection
is used for the same functionality. State of the monitored extensions is usually indicated by a series of LED lights with buttons.
Idle state – LED is off.
Ringing state – LED is blinking.
Busy or Connected state – LED is on.
The
Call Pickup
function is usually configured together with BLF allowing, for example, a receptionist to pickup ringing calls with a press of a single button.
Operator configuration
Busy Lamp Field
There is no configuration needed to make BLF work.
Call Pickup
Enable the Call Pickup feature and configure an extension (e.g. "**”). When dialing the call will be redirected to your phone.
**10 while extension 10 is ringing,
Phone configuration
Follow these steps on snom 360 (firmware 8.4.18) and snom 820 (firmware 8.4.32).
1.
Open phone administration in your browser (eg. http://192.168.1.10).
2.
Make sure that the phone is running firmware version 8.
3.
View screen
Setup > Function Keys
.
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4.
Configure BLF function on the buttons P1...Px and set: a.
First field to
Active b.
Second field to
BLF c.
Third field to
<sip:extension@ipAddress;user=phone>|callPickup
. Here extension is the monitored extension (e.g. 10).
ipAddress is the IP address of Operator (e.g. 192.168.1.1).
callPickup is the Call Pickup extension (optional).
For example,
<sip:[email protected];user=phone>|**
. In this example,
** signifies that extension
10 doesn't use the Call Pickup feature.
5.
Don't forget to save the configuration.
Call Parking
To park a call, first place the caller on hold. Then dial to a designated parking slot (e.g. *53). If you have configured BLF keys to monitor parking slots, you can simply press the key which is monitoring the parking slot (e.g. *53). After you hear the announcement you will then hear the hold music. You can then press the transfer button to join the caller into the parking slot. The caller is then parked, and you can hang up the call.
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3.1.8 How to configure Busy Lamp Field (BLF) on Well phones
Busy Lamp field (BLF) feature allows users to monitor several other extensions. Sometimes the term "Direct Station
Selection” is used for the same functionality. State of the monitored extensions is usually indicated by a series of LED lights with buttons.
1.
Idle state – LED is off
2.
Ringing state – LED is blinking
3.
Busy or Connected state – LED is on
The
Call Pickup
function is usually configured together with BLF allowing, for example, a receptionist to pickup ringing calls with a press of a single button.
Operator configuration
Busy Lamp Field
There is no configuration needed to make BLF work.
Call Pickup
Enable the Call Pickup feature and configure an extension (e.g. "**”). When dialing "**10” while extension 10 is ringing, the call will be redirected to your phone.
Phone configuration
This guide has been tested on Well SIP-T28p (firmware 2.60.9.5).
1.
Open phone administration in your browser (eg. Http://192.168.1.10).
2.
View screen
Phone > DSS Key
.
3.
Configure BLF function on the keys DSS Keys. Set:
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a.
Type to
BLF
.
b.
Value the monitored extension (e.g. 10) c.
Extension to the Call Pickup extension + extension number. (e.g. **10). This field is optional.
4.
Don't forget to save the configuration.
NOTE
It might take a while for phones to subscribe for notifications after Operator is rebooted. In this case either wait up to
10 minutes or reboot the phone.
3.1.9 Linksys/Cisco SPA: Setting the TFTP address without using the DHCP parameter 66
Use this topic, if:
You want to use
Your DHCP server does not support parameter 66.
You use the Linksys/Cisco SPA phone.
Setting the TFTP address without using the DHCP parameter 66 for Linksys SPA942
1.
In a web browser, type the IP address assigned to the phone.
2.
In the phone configuration, click
Admin Login > Advanced > Provisioning
.
3.
Enter the TFTP address in
Profile Rule
for in the following format: tftp://Kerio.Operator.IP.address/spa942$MA.cfg
4.
Click
Submit All Changes
.
5.
After the phone restarts, restart the phone manually again. In the web browser, type the following address: http://phone.IP.address/admin/reboot
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Screenshot 17: Configuring the Linksys/Cisco SPA phone
Profile rules for other Linksys/Cisco SPA telephones
Other Linksys/Cisco SPA telephones can have different administration interfaces, however, option
Profile Rule
is available in all telephones.
We prepared a list of profile rules for you:
Type
Cisco 7940,7960
Cisco 7941,7961,7960G,7940G
Cisco SPA112
Cisco SPA122
Cisco SPA301
Cisco SPA303
Cisco SPA3102
Cisco SPA501G
Cisco SPA502G
Profile Rule
N/A
N/A tftp://<ip>/_spa112<mac>.cfg
tftp://<ip>/_spa122<mac>.cfg
tftp://<ip>/Cisco/SPA301/<mac>.cfg
tftp://<ip>/Cisco/SPA303/<mac>.cfg
tftp://<ip>/_spa3102<mac>.cfg
tftp://<ip>/Cisco/SPA501G/<mac>.cfg
tftp://<ip>/spa504G<mac>.cfg
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Type
Cisco SPA504G
Cisco SPA508G
Cisco SPA509G
Cisco SPA525G
Cisco SPA525G2
Linksys PAP2T
Linksys SPA1001
Linksys SPA901
Linksys SPA921
Linksys SPA922
Linksys SPA941
Linksys SPA942
Linksys SPA962
Profile Rule
tftp://<ip>/Cisco/SPA504G/<mac>.cfg
tftp://<ip>/Cisco/SPA508G/<mac>.cfg
tftp://<ip>/Cisco/SPA509G/<mac>.cfg
tftp://<ip>/spa525G<mac>.cfg
tftp://<ip>/Cisco/SPA525G2/<mac>.cfg
tftp://<ip>/_pap2t<mac>.cfg
tftp://<ip>/Linksys/SPA1001/<mac>.cfg
tftp://<ip>/spa901<mac>.cfg
tftp://<ip>/spa921<mac>.cfg
tftp://<ip>/spa922<mac>.cfg
tftp://<ip>/spa941<mac>.cfg
tftp://<ip>/spa942<mac>.cfg
tftp://<ip>/spa962<mac>.cfg
3.2 Backups
This section provides information about server backup and data recovery.
3.2.1 Saving Kerio Operator configuration to MyKerio
3.2.2 Saving Kerio Operator configuration to FTP or local storage
3.2.1 Saving Kerio Operator configuration to MyKerio
NOTE
This information is designed for Kerio Operator 2.5 and newer.
Kerio Operator can automatically back up and upload the configuration files to MyKerio every day.
Each backup can include:
Configuration files
Local voicemail data
System logs
Call history log
License
Recorded calls
Custom provisioning files /var/tftp
To configure backup to an FTP server instead, read the
Saving configuration to FTP server
topic.
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Saving configuration to MyKerio
Before you start, connect your Kerio Operator to MyKerio. For details, read Adding Kerio Operator to MyKerio .
Once Kerio Operator is connected to MyKerio:
1.
In the administration interface, go to
Advanced Options > Backup and Recovery > Remote Backup
.
2.
In
Type
, select
MyKerio
.
3.
Select
Enable automatic daily backup
.
4.
Set the starting time and the period.
5.
Click
Apply
.
Kerio Operator uploads configuration files once a day.
Restoring configuration from a backup
To learn how to restore your configuration from a backup, read the
Backups in MyKerio topic.
3.2.2 Saving Kerio Operator configuration to FTP or local storage
Kerio Operator can back up system settings and data: to an FTP server.
to your local storage (the file can be downloaded from Kerio Operator).
to
Kerio Operator can backup the following items:
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System configuration — system settings, IVR (auto attendant scripts), users, logos, firmwares etc.
Local voicemail data — if you use
integration with Kerio Connect , Kerio Operator sends voicemails via IMAP to Kerio
Connect. These voicemails are not backed up.
SSL certificate — only an active SSL certificate is backed up.
System logs — all logs from the
Logs
section.
Call history log — all logs from the
Status > Call History
section.
License — a
.key
file with your licence.
Recorded calls — locally saved recorded calls. You can also back up recorded calls to a FTP server.
Custom provisioning files /var/tftp
Saving backups to an FTP server
1.
In the administration interface, go to section
Integration > Remote Storage
.
2.
Key in a hostname of your FTP server.
3.
Key in a username and password if it is necessary.
4.
Click
Apply
.
5.
Go to
Advanced Options > Backup and Recovery
.
6.
Change
Type
to
FTP
.
7.
Test the settings using the
Backup on Remote Storage
button.
8.
Select
Enable automatic backup to remote storage
.
9.
In the
Start at
field, specify the time at which backups should be performed.
10.
In the
Period
field, specify how often backups should be performed.
11.
Next to
Content
, click
Edit
and select content types for backup. By default, Kerio Operator backs up only a system configuration. Full backup (all items are selected) increases size of the backup.
12.
Save the settings.
Saving a single backup file
1.
Go to
Advanced Options > Backup and Recovery
.
2.
In the
Backup
section, click
Download Backup File
.
3.
Select a backup content. By default, Kerio Operator creates a full backup.
4.
Click
Create Backup for Download
.
5.
Click
Download
and save the file.
Recovering data from a backup
1.
Download a backup file from an FTP server or locally saved backup from your computer.
2.
In the administration interface, go to
Advanced Options > Backup and Recovery
.
3.
Click
Upload Backup File
.
4.
Select the file and upload the backup to Kerio Operator.
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5.
When the
Recovery
dialog box appears, select the configuration and data for recovery.
6.
Click
Recovery
.
7.
A warning about restart appears, click
OK
.
After the restart, the backup recovery is complete.
NOTE
After restoring from a backup, restart your browser in order to log back into the administration interface.
3.3 CRM integration and desktop dialers
This section helps you integrate with CRM systems and desktop dialers.
3.3.1 Salesforce integration with Kerio Operator
3.3.2 Using Kerio Operator App for Salesforce
3.3.3 Configuring OutCALL for dialing from the Microsoft Outlook contacts
3.3.4 CRM integration using the AMI
3.3.1 Salesforce integration with Kerio Operator
NOTE
Lightning Experience from Salesforce doesn't allow the settings described below. Switch to Salesforce Classic to complete the configuration.
Kerio Operator App for Salesforce
is based on Call Center. The Call Center is an application embedded in
Salesforce and integrates Salesforce with Kerio Operator. For more information about Call Centers, go to https://help.salesforce.com/
.
Kerio Operator App for Salesforce enables:
Click-to-dial.
Displaying contacts, accounts and leads during the call.
Logging calls into Salesforce.
To use Kerio Operator App for Salesforce, install the application. You can download it from Kerio Operator administration interface.
Kerio Operator supports:
Salesforce Enterprise Edition
Salesforce Performance Edition
Salesforce Unlimited Edition
This topic helps you to install and configure Kerio Operator App for Salesforce.
For more information, refer to Using
Kerio Operator App for Salesforce (page 150).
Configuring Kerio Operator
Add the Salesforce hostname to Kerio Operator and download
Call Center Definition
for Salesforce.
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1.
In the administration interface, go to
Integration
.
2.
In the
Salesforce integration
section, click
Configure
.
3.
Login to your Salesforce and copy the Salesforce hostname. Paste the hostname to
Salesforce hostname
in Kerio
Operator.
4.
Check if the Kerio Operator's hostname is complete. If the field is empty, type a correct Kerio Operator's hostname.
5.
Save the settings.
6.
Click
Download Call Center Definition
.
NOTE
The communication is based on HTTPS by default. Verify that port 443 is open in both directions and make sure that the hostname of the SSL certificate matches the Kerio Operator hostname.
For more information, refer to
Configuring SSL certificates (page 256).
Configuring salesforce.com
Configuration is divided into three steps:
Step 1: Adding Kerio Operator Call Center
To add Kerio Operator Call Center to Salesforce, follow these steps:
1.
In Salesforce, click your name and go to
Setup
.
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2.
In the
Quick Find
, type
Call Center
and click
Call Centers
in the results.
3.
Skip the help page if it appears.
4.
Click the
Import
button in the
All Call Centers
page.
5.
Click the
Choose File
button and select the
call center definition file you downloaded earlier
.
6.
Click
Import
.
Kerio Operator Call Center (Kerio Operator App for Salesforce) is installed in Salesforce. Now add users to the call center.
Step 2: Adding users to the call center
To add users (your colleagues) from Salesforce to Kerio Operator Call Center, follow these steps:
1.
In
Kerio Operator Call Center
, click
Manage Call Center Users
.
2.
Click
Add More Users
.
3.
Leave the form as it is and click
Find
.
4.
Select users and click
Add to Call Center
.
The users appear in the
Kerio Operator Call Center: Manage Users
table.
Go to
Home
in the main menu. You can see the Kerio Operator Call Center application if your user account is added in the Kerio Operator Call Center.
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Step 3: Installing the Kerio Operator Open CTI Package
Kerio Operator Open CTI Package enables searching salesforce contacts, accounts and leads in the Kerio Operator Call
Center application.
1.
Go to Salesforce.
2.
In the address bar of your browser, add this string after your Salesforce hostname (in our case it is https://eu2.salesforce.com/
): packaging/installPackage.apexp?p0=04tb0000000QG2n
... The final result is similar to: https://eu2.salesforce.com/packaging/installPackage.apexp?p0=04tb0000000QG2n
3.
A
Package Upgrade Details
page is opened.
4.
On page
Package Upgrade Details
, click
Continue
.
5.
On page
KerioOperatorOpenCti
, click
Next
.
6.
Select
Grant access to all users
and click
Next
.
7.
Click
Install
.
8.
If you are successful, the application answers that the installation is complete.
You can test all features of Kerio Operator App for Salesforce.
For more information, refer to Using Kerio Operator App for Salesforce (page 150).
Configuring number transformation for calls from Salesforce
To make calling via
Kerio Operator App for Salesforce
easy, add number transformations which ensure that numbers are dialed correctly from Salesforce.
For more information, refer to Using number transformation (page 205).
Configuring outgoing prefixes
You can also configure prefixes in Kerio Operator Call Center. However, number transformation is recommended.
1.
Go to Kerio Operator Call Center .
2.
Click
Edit
.
3.
Change prefixes in the
Dialing Options
section.
4.
Click
Save
.
Prefixes are the same for Kerio Operator and Salesforce now.
3.3.2 Using Kerio Operator App for Salesforce
Learn how to use Kerio Operator App for Salesforce. If you need to install and configure Kerio Operator App for Salesforce, go to
Salesforce integration with Kerio Operator
.
Log into Kerio Operator App for Salesforce
Dialing of numbers is available to all users who use a software or hardware phone which has an extension of the Kerio
Operator PBX configured. If the telephone is connected, it is possible to dial the called number using Kerio Operator App for Salesforce.
Before using the Kerio Operator App for Salesforce, you will be prompted to login:
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1.
Login to Salesforce, go to
Home
. You can see Kerio Operator App for Salesforce.
Screenshot 18: Kerio Operator App for Salesforce on Homepage
2.
To login to Kerio Operator App for Salesforce, type Kerio Operator credentials (the same
).
3.
Click
Login
.
If you succeed, the application is open and the extension is idle.
Screenshot 19: A successfull login with a registered extension
If your extension is offline, you have not registered phone extension. You should do the following:
If you have more extensions, change the extension (click your name and select the extension).
Check that your phone is working.
Contact your system administrator.
Dialing calls (click-to-dial) from Salesforce
To dial the number, click any number marked as a phone number.
Dialing in Salesforce works on a callback basis. This means that Kerio Operator App for Salesforce connects directly with the PBX and the PBX contacts back your phone. Therefore, the side effect of this operation is that upon clicking on
Dial
, your phone starts to ring as well as the called person's one. Pick it up and wait for the called person to answer.
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Screenshot 20: The phone number
NOTE
If the phone icon is gray, used extension is offline.
For more information, refer to Log into Kerio Operator App for Salesforce (page 150).
Displaying contact, account or lead during the call
Kerio Operator App for Salesforce can log all incoming and outgoing external calls. The call history (last five calls by default) appears directly in Kerio Operator App for Salesforce. When you click on an item connected with the phone number (it can be an account, contact or lead), the item appears and you can see the details of the caller.
Screenshot 21: Click to view detail
Call logging in Salesforce
You can log incoming and outgoing calls to the
Activity History
in Salesforce.
1.
In the Kerio Operator App for Salesforce, move your cursor onto contact, account or lead.
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Screenshot 22: Click to Log a call
2.
When a text
Click to "Log a Call"
appears, left-click. The
Log a Call
page appears.
Screenshot 23: Log a Call dialog
You can edit a task immediately.
Advanced settings
Changing prefixes
If you need to change a prefix, click your name and select
Dial out prefix
.
If you are required to dial a prefix for outgoing calls, you can configure the Kerio Operator App for Salesforce to automatically prepend dialed numbers with a prefix. If you need to assign a prefix, click your name and select dial out prefix.
Changing a language
If you need to change a language of Kerio Operator App for Salesforce, click your name and select
Change language
.
Opening Kerio Phone from Kerio Operator App
If you want to open Kerio Phone directly from Kerio Operator App, click your name and select
Open Kerio Operator
Client
.
The advantage is that you do not have to fill the credentials twice.
Configuring a time zone
Salesforce time zone should be consistent with your computer:
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1.
In Salesforce, go to
My Settings
.
Screenshot 24: My Settings
2.
Click
Personal
.
3.
Click
Language & Time Zone
.
4.
Change the time zone and save the settings.
3.3.3 Configuring OutCALL for dialing from the Microsoft Outlook contacts
OutCALL allows you to dial calls directly from Microsoft Outlook 2000 and newer. It uses the AMI interface (Asterisk
Manager Interface) which Kerio Operator supports.
Download OutCALL at http://outcall.sourceforge.net/
What you need
Microsoft Outlook 2000 or newer.
WARNING
OutCALL supports only the 32-bit version of Microsoft Outlook.
Password for dialer (AMI) generated in Kerio Operator.
Install and configure OutCALL on the user’s computer.
Settings in Kerio Operator
Read topic
for information on Kerio Connect settings. The standard settings are as follows:
1.
Login to Kerio Operator as an administrator.
2.
Open the
Configuration > Users
section.
3.
Double-click the user whom you wish to enable the OutCALL communication.
4.
This opens the
Edit user
dialog. In the event, go to the
Advanced
tab.
5.
Check
Password for dialer (AMI)
and copy the password (displayed upon clicking on the icon with keys).
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Screenshot 25: Dialer Password
Configuring OutCALL
1.
On user’s computer, install Microsoft Outlook and create a mail account (unless it has been created before).
2.
Close Microsoft Outlook before installing OutCALL.
3.
Download OutCALL at http://outcall.sourceforge.net/
.
4.
Install it on user’s computer.
5.
If the installation was successful, run OutCALL. OutCALL will run as a service with an icon displayed in the notification area (System Tray).
WARNING
For more information, refer to Troubleshooting (page 157).
6.
Right-click the icon. Context menu is displayed.
7.
Select
Settings
.
8.
This opens the
OutCALL — Settings
dialog. Go to the
Server
tab.
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Screenshot 26: OutCALL — configuring connection to Kerio Operator
9.
In the
Server
field, enter the DNS name or IP address of Kerio Operator.
10.
In the
Username
field, enter the username in the same format as it is used in Kerio Operator.
11.
In the
Password
field, enter
WARNING
does not equal the username used to login to Kerio Phone.
12.
Switch to the
Extensions
tab. The default extension is set to
10
. Click on
Edit
and enter the
(the SIP username format can be found in the
Kerio Operator
administration in section
Extensions
).
Screenshot 27: OutCALL — setting user extensions
13.
Save the settings. Once the application connects to Kerio Operator, a pop up window is displayed informing about the successful connection.
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Troubleshooting
OutCALL stops working after the start
OutCALL stops working after the start if the path to the folder contains special characters. OutCALL does not work on localized versions of Microsoft Windows XP where the Application Data folder is localized (e.g. in Czech,
"C:\Documents and Settings\user\Data aplikací"
). The problem is solved on Microsoft Vista and newer because the folder is not localized.
OutCALL stops working after the start if the username used for login to the computer contains special characters. The solution is easy: change the username so that it does not contain special characters.
Logs
If you wish to verify the communication between OutCALL and Kerio Operator, look at the logs:
Screenshot 28: The Debug log — AMI
1.
Login to Kerio Operator administration.
2.
Go to section
Logs > Debug
3.
Right-click the log area and select
Messages
.
4.
This opens the
Logging messages
dialog window. Check the
AMI (CRM Integration, Desktop Dialer Applications)
option.
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3.3.4 CRM integration using the AMI
Asterisk Manager Interface (AMI) is an interface which enables other applications to connect to Kerio Operator (which includes Asterisk) and to communicate via the AMI commands. You can use it to make phone calls. It enables you to: dial calls from your CRM system, monitor call statuses in your CRM system (e.g., create logs), direct calls to another extension or terminate calls in your CRM system.
Connecting Kerio Operator with other applications
You can connect an application with Kerio Operator very easily. The settings are different for connections with a client
(the server-to-client connection) and with a server (the server-to-server connection).
How to connect a client application (desktop application for dialing numbers) with Kerio Operator
To connect the applications, you need the username and password of the client application user:
1.
In the administration interface, go to
Configuration > Users
.
2.
Select a user and open the
Edit User
dialog.
3.
Go to tab
Advanced
and check option
Password for dialer (AMI)
.
4.
Click on the icon and note down the displayed password.
5.
Enter the username and password in the client application to authenticate.
How to connect a server (CRM system) with Kerio Operator
You need the authentication data which you enter to your CRM system:
1.
In the administration interface, go to
Configuration > Integration > General
.
2.
Click
Configure
at
Third party CTI integration (AMI)
.
3.
Check
Third party CTI integration is enabled
.
4.
Click
Add
.
5.
Enter
Account name
(usually the name of the CRM system).
6.
The password is generated automatically. Click on the icon and note down the password.
7.
To test the communication, set the permissions to full control. If the communication is successful, you may limit the permissions.
NOTE
Some applications allows you only to originate calls but they use asterisk commands which require a higher level of permission (usually full control).
8.
Login to your CRM system and enter the password for the AMI integration.
9.
Test the communication by dialing an extension.
Application we have tried and prepared a configuration guide
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What to do when communication fails
Consult the logs in Kerio Operator:
1.
In the administration interface, go to section
Logs > Debug
.
2.
Right-click on the log screen and select option
Messages
in the context menu.
3.
This opens the
Logging Messages
dialog box. Check the
AMI (CRM Integration, Desktop Dialer Applications)
.
Configure the internal firewall of Kerio Operator
1.
In the administration interface, go to section
Configuration > Network > Firewall
and check the settings.
2.
If your CRM system is located outside your local network, add its IP address in section
Configuration > Definitions >
IP Address Groups
,
3.
Go back to section
Configuration > Network > Firewall
and select a new IP address group for the integration with the CRM system.
3.4 Monitoring
This section contains information about:
3.4.1 Using Dashboard in Kerio Operator
3.4.2 Monitoring Kerio Operator
3.4.3 Managing logs in Kerio Operator
3.4.1 Using Dashboard in Kerio Operator
Kerio Operator includes a customizable Dashboard. A dashboard consists of tiles and each tile displays a different type of information (graphs, statistics, etc.). It gets displayed as you log into Kerio Operator.
It is also accessible from
Configuration > Dashboard
.
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3.4.2 Monitoring Kerio Operator
When you are experiencing problems with your connection, we recommend you to monitor the status of your PBX.
Monitoring can be done using the
Status
section:
Monitoring active calls
All current calls can be viewed under
Status > Calls
.
You can see a table where each call occupies one line and a graph displays a number of calls in time in the
Calls
section.
Go to the
Calls
section, especially in case that you plan to restart the PBX which may result in an undesired termination of a call in progress.
Call History
The Call History section keeps a list of all internal and outbound calls of the PBX.
Call History
can be viewed under
Status > Call History
.
To add or remove columns in the call history:
1.
In the administration interface, go to
Status > Call History
.
2.
Mouse-over a name of a column and click the arrow on the right side.
3.
In
Columns
, you can: select new columns to add them to the
Call History
, deselect columns to remove them.
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Each line contains information about one call. The following actions can be applied to the call history:
Action
Export to a CSV file
Clear
Description
You can click on Advanced >Export to a CSV file to save the file on your local drive.
Click on Advanced > Clear and confirm your decision in the corresponding dialog.
NOTE
Individual users can delete their history in the
Kerio Operator Softphone
data. They are not removed from the PBX and logs.
. However, this operation only hides the
Monitoring Recorded Calls
Section
Status > Recorded Calls
displays all calls recorded from
. This section displays a table where each recorded call occupies one row. Select a call to listen to it, download it to your computer or remove it.
Click
Settings
to record calls locally or to a remote storage.
For more information, refer to Saving recorded calls (page
Monitoring a Kerio Operator dial plan
A dial plan contains a list of all the used extensions and their users. You can export this list to a CSV file or print it.
Go to section
Status > Dial Plan
to see the list:
Export to CSV
— the button exports the data in the format described in table.
Extension Number
111
Type ID
1
Description
Winston Smith
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Extension Number
112
50
Type ID
1
7
Description
Ada Monroe
Voicemail
Changing the Dial Plan
If you use automatic phone provisioning and the change in your dial plan may affect automatically provisioned phones, update of the phones configuration is needed. Kerio Operator detects such changes automatically and displays a warning. If you confirm this warning, phones will be restarted at the time you selected in the dialog. You can restart the phones later manually in section
Provisioned Phones
. To restart the phones, click on the
Advanced > Restart all phones
button.
Monitoring active conferences
All current conferences can be viewed under
Status > Conferences
. The window displays two tables. Each line in the first table displays one conference. The second table displays information about individual conferences. Just select a conference and the details in the bottom table are updated.
Monitoring call queues
All active call queues and their parameters can be observed in section
Status > Call Queues
. The window displays three tables. Each line in the first table displays one call queue.
The other tables display agents and callers in a queue. Just select a queue and the details in table
Agents
and
Callers
are updated.
You can also reset the call queue statistics to start from zero. Use the
Reset
button.
System Health
The administration interface allows you to view the status of CPU, memory and disk space of your computer with Kerio
Operator.
System status can be viewed under
Status > System Health
.
In this section, click
Tasks
to: restart telephony subsystem reboot Kerio Operator power off Kerio Operator do factory reset of Kerio Operator
The
Support information
link generates an asterisk configuration file and last 100 lines of all logs. This information may be helpful especially when solving issues in cooperation with the Kerio Technologies technical support.
See detailed information about disk space usage by clicking on
Details
usage of audio files, voicemail and configuration file of Kerio Operator.
. This opens a dialog with information about disk
3.4.3 Managing logs in Kerio Operator
Logs are files where information about certain events (e.g. error and warning reports, debugging information) is recorded. Each item is represented by one row starting with a timestamp (date and time of the event). Messages in logs are displayed in English for every language version of Kerio Operator.
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Configuring logs
Logs are available in the Kerio Operator administration interface in section
Logs
.
When you right-click in a log, you can configure the following settings (available in all logs):
Option
Save log
Highlighting
Log Settings
Clear Log
Description
You can save whole logs or a selected part in a txt or HTML format. See also Log Settings option.
You can save any part of text in logs for better reference. Specify a substring or regular expression and all rows containing such text will be highlighted.
Apart from immediate savings, you can configure regular saves of individual logs, specifying the size and number of saved files. You can also enable external logging to a Syslog server.
Use this option for deleting a log.
Types of logs
Type Description
Auth
Config
Debug
The Auth log includes information about all successful attempts to login to Kerio Operator (to the administration or client interfaces).
Failed login attempts are logged into the Security log.
The Config log stores the complete history of communication between Kerio Operator Administration and the server. It is possible to determine what administration tasks were performed by a specific user.
Debug log is a special log which can be used to monitor specific information. This is especially useful for problemsolving.
To enable the Debug log, right-click in the log window and select the Messages option in the context menu. In the opened dialog window, select specific information you wish to monitor.
WARNING
In addition, displaying too much information slows Kerio Operator's performance. We recommend that you only display information that you are interested in and only when necessary.
Error
Event
Kernel
Security
Warning
The Error log displays serious errors that affect the functionality of the entire PBX. The Kerio Operator administrator should check this log regularly and try to eliminate problems found here. Otherwise, users might have problems with some services or/and serious security problems might arise.
The Event log gives information about phone and interface registrations, phone provisioning, new versions of Kerio Operator, etc.
The Kernel log contains records generated by the operating system. It includes information about starting and stopping of the server, logs generated by individual processes, etc.
The Security log contains the failed login attempts to Kerio Operator.
The Warning log shows error warnings which are not severe. Typical examples of such warnings are messages stating that a user with administrator rights has a blank password or that a user account of a given name does not exist.
Events recalling warning messages in this log do not seriously affect the PBX functionality. However, they can point at current or possible problems. The Warning log can help if for example a user is complaining that services are not working.
3.4.4 SNMP monitoring
Enabling SNMP monitoring in Kerio Operator
SNMP is a protocol which allows you to monitor Kerio Operator status.
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1.
In the administration interface, go to
Network > General
.
2.
Click
Configure
.
3.
Check
SNMP monitoring is enabled
.
4.
In the
Location
field, type any text which will help you recognize the server and its location.
5.
In the
Contact
field, type address which will help you recognize the server and its location.
6.
Select which version to use —
2c
(default password contains string public
) or
3
.
NOTE
Version 2c supports passwords as plain text only (community string), version 3 supports encryption. Some monitoring tools, however, do not support version 3.
Using Cacti for SNMP monitoring
Cacti is a standard monitoring tool which can handle the SNMP protocol.
If you use Cacti to monitor your servers, go to cacti.net
to acquire a template for Kerio Operator. It was created for Cacti 8.8
and newer and contains graphs similar to screen-shot below.
To import the template, follow these steps:
1.
Connect to your Cacti with your browser.
2.
Download the template archive and extract it.
3.
Select
Import Templates
.
4.
Click
Choose File
and select the XML template file.
Once the import is finished, go to the
Devices
section, add a new device, enter the DNS name of Kerio Operator and in the
Host Template
menu select Kerio Operator.
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3.4.5 Monitoring active calls
NOTE
This information is designed for Kerio Operator 2.3 and newer.
Call monitoring allows you to participate in any active call by dialing a special prefix, followed by an extension.
You can use call monitoring in call centers where supervisors need to monitor trainees during conversations with customers.
WARNING
When you join an active call, the active callers have no indication that you have joined the call.
Call monitoring is protected by a PIN number. Whoever knows the PIN can listen to any extension in your telephony subsystem. Therefore, we recommend to set
for people who can use the call monitoring prefix.
The default prefix for call monitoring is
*6
, and it is configured in the
PBX services . The prefix is disabled by default and
you have to enable it manually.
Configuring call monitoring
To configure the call monitoring service, follow these steps:
1.
In the administration interface, go to
PBX Services
.
2.
Double-click
Call monitoring
.
3.
In the
Edit Service
dialog, you can change the service extension.
4.
Check the
Service is enabled
option.
5.
Read the disclaimer carefully and click
I Agree
.
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6.
Click the keys icon and remember the PIN number. You can also change the PIN number. The PIN protects the call monitoring from misuse.
7.
Select the
Audio mode
:
Listen only — muted: When joining an active call in listen only mode, there is no indication to the active callers that you have joined the call.
Whisper to the extension only — muted only to remote party
Talk to both — unmuted
8.
To allow users to change the audio mode with DTMF codes, check the
Allow the monitoring user to change the mode using DTMF codes (4, 5, 6)
option. Users can change the audio mode with a key on their phone devices (
4
is for
Listen only
,
5
is for
Whisper to the extension only
and
6
is for
Talk to both
).
9.
Click
OK
.
The call monitoring service is configured.
Setting call permissions
Set a call permission group for users who can use the call monitoring feature (people who knows the PIN number): call monitoring is allowed on extensions, which can be monitored (rules 1, 2 and 3 in the figure) other calls with *6 are forbidden (rule 4 in the figure)
Example:
The first three rules allow call monitoring on extensions 111, 112, 113:
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1.
In the administration interface, go to
Definitions > Call Permission Groups
.
2.
Click
Add
.
3.
In the
Add Call Permission Group
dialog, add the name of the group.
4.
In the
Description
field, type
Group restricts call monitoring to listed extensions
.
5.
Click
Add
.
6.
In the
Add Prefix
dialog, type *6111.
7.
Switch the rule to
Allowed
and click
OK
.
8.
Repeat the steps 5, 6 and 7 for extensions 112 and 113.
The fourth rule disables general usage of *6 prefix:
1.
Click
Add
in the
Add Call Permission Group
dialog (it is still opened).
2.
In the
Add Prefix
dialog, type *6.
3.
Switch the rule to
Denied
and click
OK
.Now, you can compare your result with figure. They should be the same.
WARNING
The denial rule must be placed below the allowing rules.
4.
Click
OK
in the
Add Call Permission Group
dialog.
The group for call monitoring is established.
Now, you must assign the group to users eligible to use the call monitoring prefix and know the PIN number:
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1.
In the administration interface, go to
Configuration > Extensions
.
2.
Select an extension assigned to John Smith (in figure it is extension 201) and click the
Edit
button.
3.
In the
Edit Extension
dialog, change
Call permissions group
to
Call monitoring group
(see screenshot).
4.
Click OK.
5.
If the user has assigned more extensions, you must set
Call monitoring group
for all of them to avoid a risk of misuse of the call monitoring.
The cal monitoring group is assigned the user who is eligible to use the call monitoring prefix.
Using call monitoring
To use the call monitoring service you must know: the service extension (
*6
by default), the PIN, the monitored extension (for example
111
).
For extension
111
, dial
*6111
to listen to the conversation. Then, you will be asked for the PIN number. Now, you are silently connected to the call on extension
111
.
If you are connected to the
111
extension, you can change a mode during the call (if allowed by the call monitoring service): press
4
for listen only mode press
5
for whisper to the extension only press
6
for talk to both
You can also monitor all employees in your office: extensions in your office start with 11 five of them are assigned to employees (111, 112, 113, 114, 115)
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If you dial
*611
, you can connect to the first ongoing call from all extensions starting with 11
If you dial
*61
, you can connect to the first ongoing call from all extensions starting with 1
If you dial
*6
, you can connect to the first ongoing call from all extensions of your telephony subsystem.
Pressing
*
key will look for another call to monitor.
WARNING
As you can see, the user can monitor all calls in your telephony subsystem. Therefore, it is important to
for all users, who are eligible to use the call monitoring prefix.
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4 Settings
This section contains information about:
4.1 Phone provisioning
This section contains information about:
4.1.1 Configuring automatic phone provisioning
4.1.2 Provisioning of Kerio Operator Softphone for mobile devices
4.1.3 Accessing company contacts through LDAP on provisioned phones
4.1.4 Using provisioning tools
4.1.5 Editing provisioning templates
4.1.6 Displaying your company logo on the provisioned phones
4.1.7 How to configure phone provisioning on Polycom phones
4.1.8 Phone provisioning - wrong detection of CISCO phones
4.1.9 Uploading configuration files to Kerio Operator TFTP server
4.1.1 Configuring automatic phone provisioning
NOTE
Watch the
Configuring automatic phone provisioning in Kerio Operator video.
Phone provisioning is used for automatic configurations of selected hardware SIP phones. Phone provisioning means: phone automatically connects to the PBX after booting and is assigned a phone extension, extensions are managed in the administration interface, if you confirm or plan it, the system will perform an automatic restart of provisioned phones if needed, phone firmware is automatically updated,
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displaying a company logo on hardware phones supported by Kerio Operator
accessing company contacts through LDAP
NOTE
Automatic firmware update is not supported for the Polycom phones and the original Cisco phones (Cisco SPA is supported). However, there is a possibility to update the firmware. You can upload all necessary files to folder
/var/tftp in Kerio Operator manually.
For more information, refer to Uploading configuration files to Kerio
Operator TFTP server (page 190).
IMPORTANT
Use of phone provisioning is not always suitable. If Kerio Operator is located and runs in the Internet, for security reasons we do not recommend to use automatic phone provisioning.
What you need
1.
In your local network, you need a DHCP server supporting parameter 66 (TFTP server address). Enter the address of
Kerio Operator in this parameter.
NOTE
For more information, refer to Configuring parameter 66 in DHCP server in Kerio Control (page 268).
2.
Only selected phones support automatic phone provisioning.
3.
Appropriate settings need to be done in Kerio Operator.
NOTE
If you wish to connect a phone which is not currently supported in Kerio Operator, you cannot use automatic provisioning. The configuration must be done on the hardware phone.
How to add a phone
1.
In the administration interface, go to
Provisioned Phones > Hardware Phones
.
2.
Click
Provisioning Settings
. The configuration dialog windows is opened.
3.
Check the
Enable provisioning
option. The option must be checked.
4.
Check option
Create new extension for newly registered phones
in case you create users locally (do not map them from a directory service).
WARNING
The
Create new extension for newly registered phones
option is checked by default. If you uncheck it, you cannot use automatic remote phone restart — you will have to restart phones manually if needed.
5.
Each telephone must be authenticated when connecting to the PBX. Extension number and password are used for
SIP authentication (Master Password in this case). Option
Master password for phones is enabled
enables to create one password for all provisioned phones. The password is saved in the configuration file which is sent to the phone upon the first connection to the network and the phone will use this password to authenticate at Kerio Operator. If you disable option
Master password for phones is enabled
, all phones will have their own passwords (it can be viewed in the configuration dialog of each phone).
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Now the general environment for the provisioned phones is configured. Once a phone is connected to your network, it will be listed in section
Provisioned Phones
.
Adding phones manually
Phones which are not connected to the network can also be provisioned. You may do so manually — you need the phone's hardware address and the type of the phone. The procedure is described below:
Screenshot 29: Connecting a phone manually
1.
In section
Provisioned Phones
, click
Add
.
2.
This opens a dialog which requires the hardware address of the phone (MAC address of the network card in the phone). The address may lack the colons. Once you save it, the colons will be added automatically.
3.
Select the correct type of the hardware phone (special configuration scripts are created according to the phone type).
4.
(Optional) Set a label of the phone (for example the name of your company). The upper label on the phone display.
5.
Assign the phone user or users who will use it.
NOTE
If you do not know to which person the extension will be assigned, check option
Generate new extension number
and the extension will be assigned automatically. Phones without extensions assigned cannot be provisioned.
Importing from CSV file
Phones can be imported from a CSV file. Data in the file must follow certain rules:
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hwAddress — hardware address of the phone, phoneManufacturer — name of the phone's manufacturer, phoneType — phone type, extension1; extension2; ... — extensions assigned to the phone. The maximum number of extensions depends on the phone type.
Each phone uses one line and all items are separated by a semicolon.
The file may look as follows:
00:1a:a0:be:1e:cd;Cisco;7940;111;112
00:1b:b0:cd:e1:ca;Cisco;7960;115
00:1c:c0:ab:a2:24;Linksys;SPA942;113;114
Import data from a CSV file as described below:
1.
In the
Provisioned Phones
section, click on
Advanced > Import from a CSVfile
.
2.
This opens dialog
Import from a CSVfile
— click on
Upload CSVfile
.
3.
If the data in the file are correct, a list of all the phones and extensions is displayed. Check those you want to import.
4.
Click
OK
.
5.
The imported phones are displayed in the
Provisioned Phones
table.
Restarting provisioned phones
When you change configuration which affects provisioned phones, the phones need to be restarted (for example, when you create a new call route). When you do so, a dialog window recommending phone restart is displayed. You can do it immediately or wait for a more convenient time (for example to an off-peak time). To restart phones later:
1.
Open the
Provisioned Phones
section.
2.
Click
Advanced > Restart All Phones
.
WARNING
Some Cisco telephones from newer series are not able to restart automatically. In case of configuration changes you have to check the result. If anything is wrong, restart the phones manually.
This warning doesn't relate to Cisco SPA phones.
Firmware
Kerio Operator allows easy installation of phone firmware which are managed through the phone provisioning:
1.
Go to section
Provisioned Phones
and click on the
Advanced > Firmwares
button.
2.
In the
Firmwares and Logos
dialog, select a firmware and click
Edit
.
3.
In the
Edit firmware
dialog, select Verify the firmware. Kerio Operator vrifies if the firmware includes all important files and information.
4.
Click
Upload File
.
5.
This opens a dialog where you select a firmware file and confirm the selection.
6.
In the
New firmware
dialog, select the appropriate phone.
7.
Click
OK
.
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The new firmware is installed and after the restart will be installed to phones.
Uploading a phone provisioning module
If you want to change or create a provisioning module (archived templates + PHP scripts which can change phones behavior), download Provisioning Developer Documentation and read it carefully.
When the provisioning module is prepared and archived, upload it to Kerio Operator:
1.
Go to administration interface.
2.
In section
Provisioned Phones
, click
Advanced > Provisioning Modules.
.
3.
Click
Upload
.
4.
Restart your phones.
Overriding templates
For more information, refer to Editing provisioning templates (page 180).
What to do if you want to know the password of your phone
If any of your users needs to know the password of their phone, we do not recommend to provide them with the Master
Password. We have a specific solution:
1.
In the administration interface, go to
Provisioned Phones > Hardware Phones
.
2.
Click
Provisioning Settings
.
3.
Disable master password.
Once you disable it, each phone will have their own password which can be shared with individual users.
Configuring inter-digit timeout
Inter-digit timeout sets the time between dialing the last digit and automatic dial. If your users complains that it is too long or too short, you can adjust it:
1.
Go to the administration interface.
2.
In section
Provisioned Phones
, go to
Provisioning Settings
.
3.
In the
Phone Provisioning Settings
, set the
Inter-digit timeout
.
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How phone provisioning works
Screenshot 30: Automatic HW phone provisioning
This is how the automatic phone provisioning works:
The telephone boots in the network and sends a DHCP request for an IP address.
DHCP server accepts the request, assigns an IP address and sends it back in a DHCP reply. Besides the IP address, the message also contains TFTP (Trivial File Transfer Protocol) server address — Kerio Operator, in our case.
SIP phone connects to TFTP server integrated in Kerio Operator.
Kerio Operator checks whether the phone is new: if it is new, Kerio Operator generates a new phone extension for the phone; if it is not new, Kerio Operator finds the extension which the phone has used.
Kerio Operator generates a configuration file suitable for the particular phone type and sends it via the TFTP protocol.
The phone is configured using the values it has acquired in the configuration file and is ready to be used.
NOTE
Some phones perform an automatic restart during the configuration.
4.1.2 Provisioning of Kerio Operator Softphone for mobile devices
This topic describes auto-provisioning of Kerio Operator Softphone for mobile devices. Auto-provisioning and its functionality for SIP phones is described in
Configuring automatic phone provisioning
.
Prerequisites
Kerio Operator must have a DNS name. Type the DNS name in the
Configuration > Network
section.
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WARNING
To secure your Kerio Operator Softphones on Android devices, you must have a fully qualified domain name in the
SSL certificate of the Kerio Operator server.
Kerio Operator must use a valid SSL certificate. The certificate name must correspond with the Kerio Operator DNS name. For more information, see
Securing Kerio Operator Softphone with SSL certificates
.
Configuring provisioning for Kerio Operator Softphone
Step 1 Add a
or a
new registration of their existing extension
to users who want to use Kerio Operator Softphone.
Step 2
Step 3
Add users to provisioning:
1. In the administration interface, go to Configuration > Provisioned Phones > Softphones
2. Click Add.
3. In the Select User dialog, select the user who wants to use Kerio Operator Softphone.
4. Save the settings.
Users must configure their mobile devices to connect to Kerio Operator .
Securing Kerio Operator Softphone with SSL certificates
To secure your Kerio Operator Softphones, you must have one of the following SSL certificates:
A paid SSL certificate signed by a certification authority. These certificates do not require any further configuration.
WARNING
Do not use wildcard certificates. Kerio Operator Softphone follows the
RFC 5922 standard.
A self-signed certificate created by your Kerio Operator server. If you use a self-signed certificate, users must download and install the certificate manually. For more information, go to
Using the self-signed certificate from your Kerio
Operator server .
Configuring a dial plan
Users with Kerio Operator Softphone want to use their contact list, where phone numbers are stored in different formats.
The Dial Plan translates phone numbers from the format used in a user's contact lists to the format that can be dialed via your Kerio Operator PBX:
1.
In the administration interface, go to
Provisioned Phones > Softphones
.
2.
Click
Dial Plan Configuration
.
3.
Click
Add
to
4.
Save the rule and click
Test
in the
Dial Plan Configuration
dialog.
5.
If you need more rules, create another one.
6.
Sort rules from specific to general. Rules are applied from top to bottom.
7.
Save the settings.
Creating rules
You can use the following characters when creating new rules.
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.
Character Description
0 to 9 digits x
*#+
[ ] a single wildcard
Keyboard symbols
A collection that can include a range. For example [6-9] means 6 7 8 9. Or [136-9] means 1 3 6 7 8 9.
Repeat the last element 0 or more times. For example, with the pattern 12. the following input will match: 1 (The 2 is repeated zero times) 12, 122, 1222 and so on
Screenshot 31: Characters for your dial plan
Example 1: International calls from USA
Match number:
+x.
Remove prefix: +
Add prefix:
011
The following image describes a scenario when you wan to call from the USA (prefix
011
) to GB (prefix
+44
) and outgoing prefix of your company is
9
.
Screenshot 32: International calls from USA to GB
Example 2: Outgoing prefix 9
Match number: x.
Remove prefix: leave empty.
Add prefix:
9
Example 3: International calls in Europe (replacing + by 00)
Match number:
+x.
Remove prefix:
+
Add prefix:
00
4.1.3 Accessing company contacts through LDAP on provisioned phones
Kerio Operator offers searching in your LDAP directory from your
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WARNING
Cisco79xx phones are not supported.
Polycom phones are not supported with Kerio Connect LDAP.
Connecting to Kerio Connect LDAP/Microsoft Active Directory
1.
In the administration interface, go to
Provisioned Phones
.
2.
Click the
Provisioning Settings
button.
3.
In the
Phone Provisioning Settings
dialog, select option
Directory configuration is enabled
.
4.
Click
Configure
.
5.
Click
Configuration Wizard
.
6.
Select type of a service:
Kerio Connect LDAP — type Kerio Connect hostname, username and password.
Active Directory — type domain name and hostname of your Active Directory and credentials of account with at least read-only access to Active Directory,
Screenshot 33: The Directory Configuration dialog after finishing Kerio Connect LDAP configuration
NOTE
We recommend to create a special account with read-only access and use credentials of this account.
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3.
Save the settings.
4.
In
Provisioned Phones
, click
Advanced
and restart all provisioned phones. Phones read the new configuration and start to communicate directly with the LDAP server.
Try this feature on your phone. Find a directory on the phone and check the contact list.
For information on how to use your phone directory, read the manual of your phone.
Connecting to LDAP in general
1.
In the administration interface, go to
Provisioned Phones
.
2.
Click the
Provisioning Settings
button.
3.
In the
Phone Provisioning Settings
dialog, select option
Directory configuration is enabled
.
4.
Click
Configure
.
5.
Fill the
Directory Configuration
dialog.
6.
Save the settings.
7.
In
Provisioned Phones
, click
Advanced
and restart all provisioned phones. Phones read the new configuration and start to communicate directly with LDAP server.
Try this feature on your phone. Find a directory on the phone and check the contact list.
For information on how to use your phone directory, read the manual of your phone.
4.1.4 Using provisioning tools
NOTE
New in Kerio Operator 2.3!
Kerio Operator includes tools for phone administration. These tools can: display the phone web interface.
open a packet sniffer for a communication between the phone and Kerio Operator.
ping IP address of the phone.
Using provisioning tools
1.
In the administration interface, go to
Provisioned Phones
.
2.
Right-click a provisioned phone and in the context menu select
Tools
.
3.
Select a tool and use it.
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4.1.5 Editing provisioning templates
Kerio Operator offers overriding templates for
Template overrides were developed to allow you to change default values in the provisioning template. You can change for example remote changing BLF, speed dials or
).
Using Template Overrides
1.
Go to administration interface.
2.
In section
Provisioned Phones
, click
Advanced > Template Overrides.
.
3.
Click
Edit
.
4.
Edit the template. Kerio Operator uses the same file for a group of phones, for example for all snom phones or all Cisco
SPA phones.
5.
Click
OK
.
6.
Click
Close
.
7.
Restart your phones.
Using developer's documentation
When you edit a template, you can use input variables that come preset when your scripts are run. Variables are described in
Kerio Operator provisioning reference guide
:
1.
Download
Provisioning Developer Documentation
.
2.
Extract the ZIP file.
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3.
Open
refguide.pdf
.
You can use all variables mentioned in the downloaded document.
Example: Changing a phone's language
Lines beginning with
@ contain regular PHP code which is executed when the template is sent to the phone. So you can do things like:
@ if ($IDENT === 'snom720') { language: Deutsch
@ }
This switches the snom 720 phones' language to German.
To find out what the variable
$IDENT can contain, you can temporary add line:
@ var_dump($PHONE_TYPES);
To view the result:
1.
In the
Provisioned Phones > Hardware Phones
section, right-click the snom phone.
2.
In the context menu, click
Download Configuration
and select the correct interface.
3.
Open the saved archive and verify the variable in the file.
To differentiate between the phones, you can use variables such as
$PHONE_IP or the
$LINES array. To change a specific phone, one can do:
@ if ($PHONE_IP === '192.168.12.11') { language: Deutsch
@ }
NOTE
In this case you can not verify the functionality by downloading the file in the administration, because the
$PHONE_IP does not match.
Another example is:
@ if (isset($LINES[0]) && $LINES[0]['TELNUM'] === '21') { language: Deutsch
@ }
The example checks if the phone has at least one extension number and sets the language to German if the first extension is
21
.
4.1.6 Displaying your company logo on the provisioned phones
You can display your company logo on hardware phones supported by Kerio Operator.
What you need
Logo — each phone firmware needs a logo in a different format.
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Which type of logo do you need
1.
In the administration interface, go to
Provisioned Phones
.
2.
Click the
Advanced > Logos
button.
3.
In the
Firmwares and Logos
dialog, go to tab
Logos
.
4.
Find the firmware type installed on your phones and click
Edit
. In
Notes
, you can find the logo parameters.
Screenshot 34: Logo parameters in the Edit Logo dialog
Adding your logo to phones
The Kerio Operator logo is set by default and you have to change it:
1.
In the administration interface, go to
Provisioned Phones
.
2.
Click the
Advanced > Logos
button.
3.
Find the logo type for your phone and click
Edit
.
4.
Click
Upload File
and upload your logo.
5.
Close the dialog.
6.
In
Provisioned Phones
, click the
Provisioning Settings
button.
7.
In the
Phone Provisioning Settings
dialog, select
Display logo on the screen
.
8.
Save the settings.
9.
Restart all phones manually.
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4.1.7 How to configure phone provisioning on Polycom phones
This topic takes you through the configuration of the automatic phone provisioning via TFTP on Polycom phones.
1.
After the phone boots, click the
Menu
button.
2.
Select
Settings
and confirm.
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3.
Select
Advanced
and confirm.
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4.
Now the phone requires a password for phone provisioning.
5.
Enter the default password which is 456 (once the provisioning is configured, the password will be synchronized with the one set in the Kerio Operator administration interface in section
Provisioned Phones
).
6.
Select
Admin Settings
and confirm.
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7.
Select
Network Configuration
and confirm.
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8.
Select
Server Menu
and confirm
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9.
Select
Server Type
and use the right arrow to switch to
Trivial FTP
.
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10.
Press
Exit
to exit the
Settings
.
4.1.8 Phone provisioning - wrong detection of CISCO phones
Problem
You cannot configure a provisioned phone.
The phone has be detected and the administration interface in section
Provisioned Phones
displays a different type of phone.
The phone does not accept the extension it has been assigned.
Explanation
Some phones support two types of protocols:
SCCP
SIP
The default protocol is set to SCCP. Phones may not detect the correct communication protocol. As Kerio Operator uses
SIP for communication, the telephone cannot be attended.
How to solve it?
Switch the communication protocol of the phone from SCCP to SIP manually.
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Example: Cisco SPA 525
1.
Go to the phone configuration interface in your browser: https://phoneIPaddress/admin/advanced
2.
Change
SPA525-protocol
to
SIP
.
3.
Change
SPA525-auto-detect-sccp
to
no
.
4.
Login to Kerio Operator and go to
Provisioned Phones
.
5.
Remove the wrongly detected phone (it has been detected as an older CISCO phone).
6.
Restart the phone (so called soft restart).
Screenshot 35: Configuring the Cisco SPA 525 phone
NOTE
A similar problem has been identified for CISCO SPA 303. In this case, the phone’s administration is locked (we have found the solution on https://supportforums.cisco.com
). Further steps are similar to Cisco SPA 525.
4.1.9 Uploading configuration files to Kerio Operator TFTP server
Why to use phone or other device configuration file
phone provisioning of unsupported devices (hardware phones or other devices with a TFTP client) phone firmware upgrade
BLF configuration, ring tones (different ring tones for different phones) password change for all extension assigned to one phone
Obtaining the configuration file
The following instructions will come in handy, if you wish to change the configuration file of a provisioned phone:
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1.
In the administration interface, go to
Configuration > Provisioned Phones
.
2.
Right-click the phone whose configuration file you wish to download.
Screenshot 36: Downloading the configuration
3.
Click
Download Configuration
and select the interface. Each interface has a different configuration — different IP addresses.
4.
The ZIP file with the current configuration will be automatically saved on your computer.
Uploading new or changed configuration files to Kerio Operator
What you need
The file must be uploaded via SSH using SCP.
Locate configuration files to
/var/tftp
How to enable SSH in Kerio Operator
Follow these instructions:
1.
In the administration interface, go to section
Status > System Health
.
2.
Click
Tasks
while pressing the
Shift
key.
3.
Select
Enable SSH
.
4.
Connect to Kerio Operator via SCP (use for example WinSCP for Windows) and upload the file via SSH using SCP. For access use username root and password of a Kerio Operator administrator.
4.2 Accounts
This section helps you create user accounts and their phone extensions.
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4.2.3 Configuring multiple registration of an extension
4.2.1 Creating user accounts
User accounts in Kerio Operator are used for:
Login users to Kerio Phone
Link users with an extension
Set access rights to the system
Adding new accounts
You can create either a
or
from a
.
Adding local accounts
If you do not use directory services, create a local user in the Kerio Operator administration:
1.
In the
Configuration > Users
section, click
Add
.
2.
The
Add User
dialog box opens.
3.
On the
General
tab, type username and password. The username must not contain spaces, diacritics and special symbols.
4.
Click OK.
The user account appears in the
Users
section and the user can connect to Phone
.
Adding accounts from directory service
Mapping differs according to the directory service used:
Microsoft Active Directory
Apple Open Directory
You need basic login credentials to connect directory service to Kerio Operator.
For more information, refer to Connecting Kerio Operator to directory service (page 274).
Assigning extensions to users
An extension is an internal telephone line. Each user can have assigned one or more extensions in Kerio Operator.
1.
In the
Configuration > Users
section, select a user and click
Edit
. The
Edit User
dialog box opens.
2.
On the
Extensions
tab, click
Add
. The
Select Extensions
dialog box opens.
3.
In the
Select Extensions
dialog box, click
Add
. The
Add Extension
dialog box opens with predefined unused extension.
4.
If the extension number meets your dial plan, click OK. If not, rewrite the extension number and then click OK.
5.
Save the settings.
The users can use their Kerio Operator phone extension.
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For more information, refer to Creating extensions (page 193).
Configuring ringing rules
For more information, refer to Redirecting calls (page 209).
4.2.2 Creating extensions
An extension is an internal telephone line. Each user can have assigned one or more extensions in Kerio Operator.
The total number of extensions is limited to three times the number of licensed users.
NOTE
configured on the PBX services tab are not counted by the license file.
Adding new extensions
You have three options to add a new extension:
An extension is created automatically when you connect a
to the network.
You can create an extension in
Configuration > Users
—
the extension is assigned to a particular user
Create an extension in
Configuration > Extensions
— the extension is created as standalone (without being assigned to a user).
Creating a standalone extension
If you have a phone which is not used by any particular user, you can create a standalone extension for it.
1.
In the administration interface, go to
Configuration > Extensions
.
2.
Click
Add > Add Extension
.
3.
Type an extension number. The field suggests an unused extension. You can change the extension number manually if necessary.
4.
Save the settings.
SIP username and SIP password
Each extension has a SIP username and a SIP password. Kerio Operator uses SIP usernames and SIP passwords for authentication of phones to Kerio Operator. You use SIP username/password for connecting softphones or hardware phones to Kerio Operator.
For more information, refer to Configuring multiple registration of an extension (page
SIP usernames/passwords cannot be used to login into Kerio Operator or Kerio Phone.
Using SIP username/password
1.
In the Kerio Operator administration interface, go to
Configuration > Extensions
.
2.
Select an extension and click
Edit...
.
3.
In the
Edit Extension
dialog, you can see fields
SIP username
and
SIP password
.
4.
To display the SIP password, click the keys icon.
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Screenshot 37: SIP username and SIP password
Now you can view the SIP username/password and use it for connecting a phone to Kerio Operator.
Encrypting calls
In Kerio Operator, you can encrypt your calls for any extensions.
1.
In the Kerio Operator administration interface, go to
Configuration > Extensions
.
2.
Select an extension and click
Edit...
3.
Click the
Advanced
tab and select
Encrypt communication (TLS and SRTP)
.
4.
Click
OK
.
Now Kerio Operator encrypts all calls for the selected extension.
For more information, refer to Securing Kerio Operator (page 254).
4.2.3 Configuring multiple registration of an extension
Do you want to use your extension with various phones? Softphone in your cell phone or IP phone in your smartphone?
The solution is multiple registration.
NOTE
Multiple registration (in contrary to assigning more extensions to one user) gives user the possibility to call from the same extension any time they make a call.
EXAMPLE
User Brenda Roar with username broar working at the Marketing department uses the extension 224. When necessary, she also works from home. She uses the following to communicate:
1.
She has an automatically provisioned phone Cisco 7940 in his office.
2.
She has X-Lite softphone on her home computer.
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3.
Occasionally, when connected via WiFi, she uses a SIP client on her mobile phone.
With correct settings of multiple registration that will be described in the following chapter she can use all the beforementioned methods to authenticate.
Creating multiple registrations
1.
Open section
Configuration > Extensions
.
2.
Select Brenda Roar’s extension (224). Click on
Add > Add Another Registration
.
Screenshot 38: Extensions > Add Another Registration
3.
A new registration is added to the user table. Add another registration. The result should be similar to the following image.
Screenshot 39: Extensions > Multiple registration
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4.
Double-click the
224p1 registration and note the SIP username and SIP password from the opened dialog.
Screenshot 40: Edit Extension > Login information for X-Lite
5.
Click OK.
6.
In the X-Lite settings (detailed information for installation can be found in topic
Configuring the X-Lite software phone ), enter the newly generated string into
User ID
and the SIP password into
Password
.
7.
Repeat steps 4 to 6 for the second registration for the SIP client on a mobile phone.
4.3 Numbering
This section provides information about:
4.3.1 Mapping external and internal numbers
4.3.2 Displaying, hiding and overriding phone numbers
4.3.3 Setting emergency numbers
4.3.4 Using number transformation
4.3.5 Adding area codes to called numbers
4.3.1 Mapping external and internal numbers
In Kerio Operator you can map external numbers to internal extensions. You can:
Strip the first
0-n digits from the number, including reducing the number to an empty string
Add other digits to the beginning of the number
Routing incoming calls
In Kerio Operator, you can use rewriting rules to map numbers for SIP and standard phone interfaces. Depending on your provider's requirements, you may need to strip out or change numbers
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Example:
A company has 100 phone numbers from a telephone provider.
For incoming calls, the provider sends the whole number.
For outgoing calls, the provider requires the whole number.
Internal extensions have the format
2xx
.
The prefix for outgoing calls is
9
.
When external
Phone A
(with the number
5550399
) calls internal
Phone B
(with the number
5550101
and the internal extension
201
):
1. Phone A dials
Phone B's
number and a signal goes to the provider.
2.
The provider sends the number to Kerio Operator.
3.
The rewriting rule strips five digits from the left and adds the prefix
2
.
4.
The call connects.
Mapping a trunk of numbers
To set the interface for an interval of numbers (
55501xx in this example):
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of outgoing calls
.
2.
Select the routing rule for the provider interface and click
Edit
. The
Edit Incoming Call
dialog box opens.
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3.
In the
Called number
section, strip the first five digits from the left, and add the prefix
2
. This modifies the number to the final format of the extension (
2xx
).
4.
In the
Calling number
section, do not strip out any digits, and add the prefix
9
. This is useful when you want to call back the external number.
5.
Click
OK
.
Mapping a single number or multiple numbers
To set the interface for single or multiple numbers (
5550100 to 5550199 in this example):
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select the routing rule for the provider interface and click
Edit
. The
Edit Incoming Call
dialog box opens.
3.
Double-click a line in the
Extension
column and assign an extension to the external number.
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4.
In the
Called number
section, strip the first two digits from the left, and add the prefix
2
. This modifies the number to the final format of the extension (
2xx
).
5.
Click
OK
.
Routing outgoing calls
You can configure outgoing calls when creating an interface, either
or
.
For rewriting the numbers, you need additional configuration.
Example:
External
Phone A
has the number
5550199
.
Internal
Phone B
has the number
5550101
and the internal extension
201
.
For outgoing calls, Kerio Operator uses the prefix
9
.
The provider needs the whole number for outgoing calls.
When
Phone B
calls
Phone A
:
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1. Phone B dials the number with the
9
prefix (
95550199
).
2.
Kerio Operator uses rewriting rules and strips out the first digit (
9
). The number Kerio Operator sends to the provider is
5550199
.
3.
The provider connects to
Phone A
.
To achieve this configuration:
1.
In the administration interface, go to
Configuration > Call Routing > Routing of outgoing calls
.
2.
Select an interface and click
Edit
. The
Edit Outgoing Route
dialog box opens.
3.
In the
Called number
section, strip one digit from left and do not add a prefix.
4.
Click
OK
.
Rules for outgoing calls
You can configure rules for outgoing calls:
1.
In the administration interface, go to
Configuration > Call Routing > Routing of outgoing calls
.
2.
Select an interface and click
Edit
.
3.
In the
Calling number (Caller ID)
section, select one of these options:
Map extensions to external numbers based on routing of incoming calls
if you want to use a table of external numbers configured for the provider
Assign the default number to all extensions
if you want to use a default number for all extensions
Rewrite extension numbers (default number not used)
if you want to rewrite numbers in a specific way
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Exceptions to the outgoing routes
To create an exception:
1.
In the administration interface, go to
Configuration > Call Routing > Routing of outgoing calls
.
2.
Select an interface and click
Edit
.
3.
Enable the
Use route only for numbers defined in exceptions
option.
4.
Click the
Exceptions
tab and click
Add
.
5.
To change the internal number, double-click the displayed extension and select a new extension.
6.
To change the external number, double-click the displayed number and select a new number.
7.
If you want to hide this extension's number so the call recipient cannot see it, select the box in the
Hide Caller ID
column (see
Displaying, hiding and overriding phone numbers
for more details).
8.
Click
OK
.
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Working with prefixes for outgoing calls
Kerio Operator works with prefixes for outgoing calls in a specific schema and you can use one prefix for multiple providers. Kerio Operator uses the longest prefix matching the dialed number. If that dial attempt fails, Kerio Operator tries the next route with the same prefix.
Example
Use the prefix
011
for two providers (
provider1
and
provider2
) and the prefix
0
for outgoing calls.
Dials the number
011 234 567
.
After dialing this number:
1.
Kerio Operator goes through the
Routing of outgoing calls
table and tries to match the prefix.
2.
Kerio Operator finds two matching prefixes,
0
and
011
, and uses the longest prefix.
3.
Kerio Operator tries the
011
prefix to connect to
provider2
.
4.
If the connection does not work, Kerio Operator uses the same prefix to connect to
provider1
.
5.
If the connection still does not work, Kerio Operator does not try to use the last prefix (in this example, the
0
prefix), and the call fails.
Changing the order of prefixes
Kerio Operator works with providers for the same prefix in order from top to bottom. You can change that order by the using arrows on the right side of the administration interface to move it up or down.
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4.3.2 Displaying, hiding and overriding phone numbers
Hiding users´ phone number
NOTE
Redesigned in Kerio Operator 2.4!
To hide users´ phone numbers for outgoing calls:
1.
In the administration interface, go to the
Configuration > Call Routing > Routing of outgoing calls
section, select a prefix and click
Edit
. The
Edit Outgoing Route
dialog box opens.
2.
Go to the
Exceptions
tab.
3. Add an extension.
4.
Select the box in the
Hide Caller ID
column.
5.
Click
OK
.
NOTE
Some VoIP service providers do not allow hiding of phone numbers. If you use one of these providers, this settings do not work. See topic
Connecting to VoIP service provider
Changing phone number to a name
For outgoing calls, you can change the phone number to display a name:
1.
In the administration interface, go to the
Configuration > Call Routing > Interfaces and routing of incoming calls
section, select an interface and click
Edit
. The
Edit External Interface
dialog box opens.
2.
Go to the
Advanced
tab.
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3.
In the
Outgoing calls
section, select the
Override display name with
option, and type a new name.
4.
Click
OK
.
Extending display names for incoming calls
NOTE
New in Kerio Operator 2.4!
In Kerio Operator, you can extend the display name of incoming calls. The configuration works for all numbers that reach the interface and Kerio Operator adds the configured text to the beginning of the number or the caller's ID.
For example, a call center provides a technical support for several companies (for example,
Workplace
). Administrator wants to extend a display name of incoming calls with the company name, so the call center employees know from where comes the call:
1.
In the administration interface, go to the
Configuration > Call Routing > Interfaces and routing of incoming calls
section.
2.
Select an interface and click
Edit
. The
Edit External Interface
dialog box opens.
3.
Go to the
Advanced
tab.
4.
In the
Incoming calls
section, select the
Prepend display name with
option, and type
Workplace -
.
5.
Click
OK
.
After this configuration, Kerio Operator extends all incoming calls to this interface with
Workplace -
(for example,
Workplace - 555 0155
).
4.3.3 Setting emergency numbers
When configuring emergency numbers, you can: add emergency numbers to the system, enable direct dialing (without the prefix for calling external networks).
NOTE
and
are not applied to emergency numbers.
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Configuring emergency numbers
1.
In the administration interface, go to
Configuration > Call Routing
.
2.
Click the
Emergency Numbers
button placed in the lower left corner.
3.
Click
Overwrite
and select the country.
4.
If the lists of emergency numbers do not suit your needs, click
Add
to create your own emergency numbers.
Enabling direct dialing
All outgoing calls to external networks use a prefix. You can configure an exception for emergency numbers:
1.
In the administration interface, go to
Configuration > Call Routing
.
2.
Click the
Emergency Numbers
button placed in the lower left corner.
3.
Check
Enable direct dialing
.
4.
Select
Used outgoing route
. This route will be used for all calls to the emergency numbers.
WARNING
If the direct dialing is enabled, you cannot create extensions which equal the emergency numbers.
4.3.4 Using number transformation
A number transformation in Kerio Operator ensures that phone numbers dialed automatically by an application (such as
Click to Call ) are dialed in the right format. The right format is the same format as for usual calls — without the outgoing prefix, or without the international call prefix. It depends on your SIP provider and their SIP server settings.
You may need the number transformation if your users use:
Click to Call in Kerio Connect Client
Kerio Operator App for Salesforce
Kerio Phone
All the above mentioned applications dial phone numbers in the same format as they are displayed. If the number has an international prefix, Kerio Operator must delete it. If your Kerio Operator uses a prefix for outgoing calls, you must create a rule for adding the prefix in front of the phone number.
Configuring a number transformation
If you use an outgoing prefix in your environment, you must add a number transformation rule to Kerio Operator:
1.
In the administration interface, go to
Integration
.
2.
On the
Number Transformation
tab, add the rule for your outgoing prefix (for example 9).
3.
Click
Add
.
4.
In the
Add Number Transformation Rule
dialog, type dot in the
Match number
field. Numbers of any length are matched.
5.
In the
Add prefix
field, add the outgoing prefix (for example 9).
6.
Click
OK
.
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Example
The example uses the US international prefix and shows you a number transformation if:
9 is an outgoing prefix.
Your SIP provider cannot dial numbers starting with a +.
In case of local calls, you want to strip the international prefix.
In case of international calls, you want to change + to 011.
If the number does not start with 9, the rule adds 9 in front of the phone number.
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4.3.5 Adding area codes to called numbers
In some situations you need to add an area code to your dialed numbers. In Kerio Operator, you can set the area code automatically.
Example:
You use only 7-digit schema for your phone numbers (for example 555-5555)
Your provider accepts only 10-digit numbers
To change your schema from 7-digit to 10-digit numbers:
1.
Go to
Configuration > Call Routing
.
2.
Click
Add...
under the
Routing of outgoing calls
section.
3.
On the
General
tab, add prefix number (for example 9).
4.
Select your interface.
5.
For
Called Numbers
set
Strip digits from left
to 0 and type a 3-digit number prefix (for example 450).
6.
Click
OK
.
After that, all outgoing calls dialed with the 9 prefix have 10-digit format (in our example 450555-5555 instead of 555-
5555).
Disabling outgoing calls to certain countries or regions
For more information, refer to Disabling outgoing calls to certain countries or regions (page 216).
4.4 Call settings
This section provides information about:
4.4.1 Bandwidth used by the different codecs 208
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4.4.2 Using Opus codec for Kerio Phone
4.4.4 Blocking incoming calls in Kerio Operator
4.4.5 Disabling computer calls for Kerio Phone
4.4.6 Disabling outgoing calls to certain countries or regions
4.4.7 Video calling in Kerio Operator
4.4.1 Bandwidth used by the different codecs
When you are using Voice over IP, the used VoIP phones and PBX can use different so called codecs. The consumption of bandwidth depends on which codec you use.
The following table gives you a short overview of the bandwidth consumption of the different codecs. On the page from asteriskguru.com, you can also find a bandwidth calculator tool .
codec
G.711 a/u-law
G.729
iLBC
G.723.1
G.723
GSM fullrate
G.726
SpeeX
name
PCM
CS-CELP iLBC
MP-MLQ
A-CELP
bandwidth (incl. overhead)
80 kBit/s
32 kBit/s
32 kBit/s
21 kBit/s
15 kBit/s
GSM 13 kBit/s
AD-PCM 55 kBit/s
SpeeX 4 - 15 kBit/s
bandwidth for 5 concurrent calls
512 kBit/s
200 kBit/s
200 kBit/s
110 kBit/s
80 kBit/s
80 kBit/s
386 kBit/s
25 - 80 kBit/s
quality
ISDN good good average average average
GSM variable
4.4.2 Using Opus codec for Kerio Phone
NOTE
New in Kerio Operator 2.5!
Kerio Operator allows you to use the Opus codec for calls via Kerio Phone for desktop and web. To use Opus for all your calls:
1.
In the Kerio Operator administration interface, go to
Advanced Options > Telephony
.
2.
In the
Codec configuration
section, select the
Prefer Opus codec
option.
3.
Click
Apply
.
NOTE
Kerio Operator transcodes Opus to another codec every time the other caller doesn't use it. Transcoding calls increases the CPU usage. If you expect larger amount of concurrent calls, disable this option.
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4.4.3 Redirecting calls
NOTE
Redesigned in Kerio Operator 2.4!
Kerio Operator can route incoming calls to different internal extensions or external numbers.
You can configure ringing rules (call forwarding) for each user in the
Ringing Rules
section.
NOTE
Users can also change their ringing rules in the Phone interface in the
Forwarding
section.
Configuring ringing rules in the administration
See the following example:
Bob has the internal extension
11 and a cell phone with the number
5550155
. He wants to receive calls on his cell phone. When he is busy, calls fallback to voicemail.
1.
In the administration interface, go to
Configuration > Users
.
2.
Select an account and click
Edit
. The
Edit User
dialog box opens.
3.
Go to the
Extensions
tab.
4.
Select an extension and click
Ringing Rules
.
5.
Enable the
Ring extension
option.
6.
Select a number for
Timeout
. When the specified time runs out, Kerio Operator forwards the call.
7.
For
When busy
, select the
Continue
option.
8.
Click Add and type the number
5550155
and a description (cell phone).
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9.
Select a number for
Timeout
.
10.
Enable the
Fallback to voicemail
option.
11.
Click
OK
to save your changes.
Additional configuration
NOTE
New in Kerio Operator 2.4!
For ringing rules, you can configure additional settings:
Configure extension to allow only one incoming call
Apply ringing rules to calls coming from call queues and ringing groups
Configuring extensions to allow only one incoming call
If your phones support multiple calls, you can configure your extensions to reject or redirect additional incoming calls when an extension is already busy with a call.
To allow only one incoming call at a time:
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1.
In the administration interface, go to
Configuration > Users
.
2.
Select an account and click
Edit
. The
Edit User
dialog box opens.
3.
Go to the
Extensions
tab.
4.
Select an extension and click
Ringing Rules
.
5.
Enable the
Allow only one incoming call
option.
6.
Click
OK
.
Kerio Operator now handles incoming calls using the configuration set in the
Ringing Rules
dialog box.
Applying ringing rules to calls coming from call queues and ringing groups
To configure ringing rules for calls from call queues and ringing groups:
1.
In the administration interface, go to
Configuration > Users
.
2.
Select an account and click
Edit
. The
Edit User
dialog box opens.
3.
Go to the
Extensions
tab.
4.
Select an extension and click
Ringing Rules
.
5.
Enable the
Use the above rules also for Ringing Groups and Call Queues
option.
6.
Click
OK
.
Configuring call forwarding in Kerio Phone
For more information refer to Redirecting calls in Kerio Phone .
4.4.4 Blocking incoming calls in Kerio Operator
If you want to block incoming calls from certain numbers, you can add the numbers to Kerio Operator's
Blacklist
.
Kerio Operator then blocks all numbers in the blacklist. No incoming calls from these numbers are connected.
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Adding numbers to the blacklist
1.
In the administration interface, go to
Configuration > Blacklist
.
2.
Click
Add
.
3.
Type the number you want to block (
Match number
). You can match an entire number, or you can use
X for single characters and
.
(dot) for multiple characters.
4.
Add a description to document the reason for blacklisting the number.
5.
Click
OK
.
6.
Add as many rules as you need.
7.
(Optional) You can also block anonymous callers.
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NOTE
Do not use this option if your provider does not show the caller's number. Otherwise, all incoming calls are blocked.
8.
Click
Apply
.
When you receive a call from any of the numbers in the blacklist, your extension appears to be busy and the call is not connected.
Adding numbers from Call History
In
Call History
, you can select any incoming call number to add to the blacklist.
Right-click the number (a line) and select
Blacklist
.
When a call is blocked by blacklisting, you see
Blacklisted
in the
Status
column in
Call History
.
Adding/removing numbers with a PBX service
You can also use your phone to add numbers to the blacklist.
Kerio Operator has three pre-defined PBX services:
*30 for adding numbers to the blacklist
*31 for removing numbers from the blacklist
*32 for adding the last caller to the blacklist
To add a number to the blacklist:
1.
Dial the service number for adding numbers:
*30
.
2.
After the beep, enter the phone number.
3.
Hang up.
To add the last caller to the blacklist:
1.
Dial the service number for adding last number:
*32
.
2.
Confirm the number.
3.
Hang up.
To remove a number form the blacklist:
1.
Dial the service number for removing numbers:
*31
.
2.
After the beep, enter the phone number.
3.
Hang up.
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Importing blacklists
You can prepare a
CSV file of numbers to be blocked and import init to Kerio Operator.
Each line in the file defines one entry. Entries must have the following format:
0,"555.","All numbers beginning with 555 are blocked"
1,"+.","All calls from foreign countries beginning with +"
1,"00.","All calls from foreign countries beginning with 00"
1,"X906.",""
NOTE
Separe all items with commas, and put number definitions and descriptions inside quotation marks.
If any item is empty, keep the quotation marks.
To import the file, go to the
Blacklist
section and click
Import from a CSVfile
.
Exporting blacklists
You can export the list of blacklisted numbers to a
*.csv
file.
1.
Click
Export to a CSVfile
.
2.
Go to the correct folder, assign a file name, and save.
4.4.5 Disabling computer calls for Kerio Phone
NOTE
New in Kerio Operator 2.5.2!
Kerio Operator enables you to make calls via Kerio Phone for desktop and web using WebRTC.
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You can disable the WebRTC support in the administration, so that users cannot see and use the
Computer
extension in their applications.
You can disable computer calls for:
A single user
Multiple users
All users on your Kerio Operator Server
Disabling computer calls for a single user
1.
In the administration interface, go to
Configuration > Users
.
2.
Select a user and click
Edit
.
3.
Switch to the
Advanced
tab.
4.
Deselect the
Computer calls from Kerio Phone
option.
5.
Click
OK
.
At this point, a selected user can no longer use the
Computer
extension for making calls.
Disabling computer calls for multiple users
1.
In the administration interface, go to
Configuration > Users
.
2.
Select multiple users and click Edit.
3.
Switch to the
Advanced
tab.
4.
In the
Computer calls
field, select
No
.
5.
Click
OK
.
At this point, selected users can no longer use the
Computer
extension for making calls.
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Disabling calls for all users
1.
In the administration interface, go to
Configuration > Advanced Options
.
2.
Switch to the
Telephony
tab.
3.
In the
Kerio Phone for desktop and web
section, deselect the
Enable computer calls
option.
4.
Click
Apply
.
At this point, users on your Kerio Operator server can no longer use the
Computer
extension for making calls.
4.4.6 Disabling outgoing calls to certain countries or regions
For security reasons, disable calls to countries users never call, create call permission groups and assign them to extensions.
Call permission groups can:
Allow everything and disable certain prefixes, or
Disable everything and allow certain prefixes
Disabling outgoing calls
1.
In the administration interface, go to
Definitions > Call Permission Groups
.
2.
Click
Add
or select an existing group and click
Duplicate
.
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3.
In the
Add Call Permission Group
dialog box, type the name and a description for the group and click
Add
.
4.
Type a specific string of numbers, and choose the option to allow or deny access.
WARNING
To limit outgoing calls, include the prefix for outbound calls (usually 9).
5.
(Optional) Repeat steps 2 and 3 to add additional numbers.
6.
Click
OK
to save the settings.
NOTE
Kerio Operator applies the calls permissions in order, one by one.
Assigning call permission groups to extensions
1.
In the administration interface, go to
Configuration > Extensions
and assign the created call permission groups to individual extensions.
2.
Select an extension and click
Edit
. The
Edit Extension
dialog box opens.
3.
Select a
Call permissions group
.
4.
Click
OK
.
NOTE
To assign a call permission group for multiple extensions, select multiple extensions and click
Edit
.
Adding area codes to called numbers
For more information, refer to Adding area codes to called numbers (page 207).
4.4.7 Video calling in Kerio Operator
NOTE
New in Kerio Operator 2.4!
Kerio Operator now supports video calls with video enabled devices or software.
Prerequisites:
Devices or software that use the same supported video codecs
Configured extensions and interfaces to use the same video codec as your devices
Kerio Operator supports these video codecs (all are pass-through only):
H.261 Video
H.263 Video
H.263+ Video
H.264
MPEG4 Video
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Adding video codecs to extensions
To enable video codecs for any extension:
1.
In the administration interface, go to
Configuration > Extensions
.
2.
Select an extension and click
Edit
. The
Edit Extension
dialog box opens.
3.
Go to the
Codecs
tab.
4.
Select a codec and click
Add
to insert the codec in the
Selected codecs
list.
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5.
Click
OK
to save your settings.
WARNING
All extensions participating in a video call must have the same codec.
You can select a single codec and assign it to all your extensions.
Adding video codecs to interfaces
To enable video codecs for any interface:
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select an interface and click
Edit
. The
Edit External Interface
dialog box opens.
3.
Go to the
Codecs
tab.
4.
Select a codec and click
Add
to insert the codec in the
Selected codecs
list.
5.
Click
OK
to save your settings.
WARNING
Interfaces must have the same codecs as all extensions participating in a video call.
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Troubleshooting
Video codecs in Kerio Operator are pass-through only and Kerio Operator cannot transcode them. For a proper connection, all devices must use the same codec. See the examples below:
Example of improper configuration
Device A
tries to manage a video call with
Device B
:
Device A
works with the
H.261 Video
codec.
Device B
works with the
H.263+ Video
codec.
This configuration does not work, because the devices have different codecs and Kerio Operator cannot transcode them.
Example of proper configuration
Device A
tries to manage a video call with
Device B
:
Device A
works with the
H.264 Video
codec.
Device B
also works with the
H.264 Video
codec.
This configuration works, because both devices work with the same codec, so Kerio Operator does not need to transcode any codecs.
Phones do not display any video
If your phone does not display any video during the call:
Set the same codecs for each device. To verify which codecs devices use, see
Lower the resolution on the caller's phone.
For example,
Grandstream GXV3272
sends video call with 720p resolution to
Grandstream GXV3140
, but
Grandstream
GXV3140
cannot decode the video. User decreases the resolution on
Grandstream GXV3272
and both phones start to display the video.
Phones do not transmit video
If your phone does not transmit video call, configure the device to make a video call.
For example, before you make the call, configure
Yealink VP-530
to prefer video calls.
Video is unstable
Devices with slow CPU or without a hardware acceleration can have problems with decoding the video:
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Decrease the resolution on the caller's phone.
Verify that the network is not jammed. For example, transmitting a VGA signal using a
H.264 codec
takes
400 kbps in each direction.
4.5 PBX services
This section contains information about configuring and using PBX services.
4.5.2 Configuring music on hold
4.5.4 Configuring and using call parking
4.5.5 Configuring and using conferences
4.5.6 Configuring auto attendant scripts
4.5.7 Setting time conditions in auto attendant scripts
4.5.8 Using the Day/night mode in auto attendant scripts
4.5.10 Configuring call queues
4.5.1 Using PBX services
Kerio Operator has special phone extensions which run the following services:
— a service extension to access voicemail. Kerio Operator recognizes which extension is used and you can set if PIN is required or not. This service is set automatically for provisioned phones.
Voicemail with login prompt
—a service extension to access voicemail. Kerio Operator is not able to recognize which extension is used. Users must authenticate with typing their extension and PIN.
Echo
— this option helps you monitor whether phones are correctly connected and what is the sound delay. Speak to the phone after hearing the automated message. If done correctly, your message is recorded and played back.
Music
— music plays upon dialing the extension (used for checking the connection).
Current time
— auto attendant tells the current date and time.
Dial by extension
— auto attendant invites the user to enter the extension which the operator will dial.
Dial by name
— user enters first several letters of the callee's surname and system searches among the users created in Kerio Operator and dials the extension.
Record audio
— Kerio Operator starts recording. Thus you can easily
for auto attendant scripts in excellent quality.
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Receive fax messages
— the service enables you to receive fax to defined email address. Necessary condition for enabeling the service is entering email address for receiving faxes in PDF format.
To configure PBX services, go to the administration interface > PBX Services.
WARNING
If you wish to use any service, tick the box next to this service. Extensions offering the services are disabled by default.
Creating voice files
This chapter shows how to create a records for an auto attendant script easily, fast and in sufficient quality.
1.
Prepare texts.
2.
In the administration interface, go to
PBX Services
, enable
Record audio
and save the settings.
3.
Pick up the handset of your phone which is connected to Kerio Operator.
4.
Dial the
Record audio
service.
5.
Say individual voice recordings into the headset.
The record is stored in the audio file library in Kerio Operator. You can listen and manage the recordings in the
Definitions > Audio File Library
.
NOTE
If you open the administration interface in Safari browser and you cannot play any recordings, read topic
Cannot play voicemails or audio files in Safari
4.5.2 Configuring music on hold
While a caller is waiting for connection or in a call queue (see the
topic), they can hear recorded music. Kerio Operator has a default music collection. You can add and configure other audio files. You can upload any file in GSM and WAV format in section
Definitions > Music On Hold
.
Adding new collections
To add a new music collection (with one or more file), follow these instructions:
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Screenshot 41: Adding New Collection
1.
Go to
Definitions > Music On Hold
and click the
Add
button.
2.
In the
Add Music on Hold Collection
, enter a name for the collection and a description.
3.
Click the
Add
button situated on the right side of the table with added audio files.
4.
In the
Select Audio File
dialog, add file one by one by clicking
Upload
.
5.
Select a file in the list and double-click it. Repeat this step until all your uploaded files are listed in table
Audio files in the collection
.
Setting Default Collection
In the
Add Music on Hold Collection
dialog, check the
Make this collection the default music on hold
to ensure this collection is used as default in all other Kerio Operator Administration settings.
The default collection is used while holding the line (usually the
Hold
button on most phones). The other collections can be used, for example, in call queues.
4.5.3 Configuring voicemail
Voicemail does not need any configuration. It works automatically once Kerio Operator starts. All users have forwarding to voicemail inbox enabled by default: when unavailable when busy
You can change the settings in section
Users
(
). Users can also modify the settings in their Kerio Phone.
You can find the advanced voicemail configuration in the administration interface in section
Voicemail
.
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Screenshot 42: Configuration > Voicemail
What is direct access to voicemail inbox and how to configure it
Direct access to users' voicemail enables the receptionist to connect calls directly to callee's voicemail.
1.
In the administration interface, go to
Voicemail > General
.
2.
Check
Allow direct dialing to user's voicemail boxes
.
3.
Type a prefix in
Prefix for direct dialing
.
4.
(Optional) Set an announcement (greeting message). If a call is redirected to voicemail, the caller hears a recorded message. This message can consist of two parts:
Instructions
inform callers what they should do next: Leave a message after the beep.
Message
informs callers that the callee is unavailable (the phone is switched off) or busy (the callee speaks with someone else).
5.
(Optional) To change the size of users' voicemail boxes, adjust the value in
Maximum messages in each voicemail box
.
6.
(Optional) To automatically delete read messages in full voicemail boxes, select
Automatically delete the oldest read message if the voicemail box is full
.
7.
Click
Apply
.
Now the receptionist can dial the extension for direct access followed by the user's extension. The caller will be directed to the voicemail box of the person they are calling.
Enabling caller to escape voicemail by dialing 0
If you want to enable escaping voicemail by dialing 0, you must set an extension where the call is redirected:
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1.
In the administration interface, go to
Voicemail > General
.
2.
Select
After pressing the 0 key, dial extension
.
3.
Type an extension.
4.
Click
Apply
.
Configuring forwarding of voicemail messages to user’s email inbox
To send voicemail messages to email inboxes of the users, you need to set their email addresses in the administration interface in
Users
.
NOTE
If the users' INBOXes are unavailable (the mailserver is down), the user accounts are disconnected from voicemail and try to reconnect every 5 minutes. Each attempt to connect is recorded in logs.
My mailserver is Kerio Connect
For more information, refer to Integrating Kerio Connect and Kerio Operator (page 295).
My mailserver is a different SMTP server
1.
On your mail server, create a special user which will be used for sending the voicemail messages. You can name them for example operator
.
2.
Go to administration interface to
Voicemail > tab Email
and check
Send each message to user's email
.
3.
In
Mail server hostname
, type the SMTP server hostname and click
SMTP Configuration
.
4.
Set the port number of the port used by your SMTP server. Usually 25 for SMTP and 465 for SMTPS
5.
Decide, whether to communicate through secured connection. If the configuration of your mail server allows it, we recommend the encrypted connection to establish more secure communication.
6.
If your SMTP server requires authentication, check
Server requires authentication
. Use the username and password for the account you created on your mail server in step 1.
7.
Click
OK
.
8.
In
Voicemail > tab Email
, type a valid email address in
Sender email address
(so that your antispam rules accept it).
The address should also represent the origin of the message. Example: [email protected]
Configuring the welcome message for callers
If a call is redirected to voicemail, the caller hears a recorded message. This message can consist of two parts:
Instructions
inform callers what they should do next: Leave a message after the beep.
Message
informs callers that the callee is unavailable.
How to set the greeting message?
1.
Open section
Voicemail
.
2.
In the
Greeting message
menu, select whether the caller will hear the instruction, the message or both.
Greeting message for the direct dialing is described in the
What is direct access to voicemail inbox and how to configure it
section.
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Changing the extension and voicemail PIN
Users use a special extension number to access their voicemail (by default: 50 or 51) and PIN.
To change the extension or enable/disable PIN, go to section
PBX Services
and read topic
Screenshot 43: PBX Services
To set the user’s PIN, go to account configuration in section
Users
to tab
Extensions
.
Accessing voicemail
On your phone, press voicemail button or dial voicemail number and play the message.
Through Kerio Phone.
WARNING
For users of Apple iPhone, iPad or Apple Mac OS X: If you cannot play your voicemail messages in Kerio Phone, contact the Kerio Operator administrator.
An invalid certificate may be the reason
.
By forwarding voicemail to your mailbox (to get more information on this option, contact your network administrator).
Removing voicemail data for selected user
You can remove all local data connected with the particular user.
Local data is: voicemail custom voicemail greeting message
NOTE
Local data means that you cannot use this feature when you use the Kerio Connect integration — voice messages are stored in Kerio Connect in this case.
1.
In the administration interface, go to
Users
.
2.
Right-click the table heading.
3.
In the context menu, select
Columns > Voicemail
and
Columns > Local Voicemail Size
.
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Screenshot 44: Table context menu
4.
Right-click the selected user and click
Erase Local Voicemail Data
.
Screenshot 45: User's context menu
If you succeed, there is value
0B in the
Local Voicemail Size
column.
Managing voicemail via Kerio Phone
For more information refer to Using Kerio Phone
4.5.4 Configuring and using call parking
Call parking is a special type of call transfers. Parked calls wait for the callee on
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Configuring call parking
You can park calls on numbers which consist of:
PBX service prefix
Parking position number
1.
In the administration interface, go to the
PBX Services
section.
2.
Double-click
Call parking
to open the
Edit Service
dialog box.
3.
Select the
Service is enabled
option.
4.
In the
Service extension
field, type the call parking prefix. You can leave the default prefix setting
*5
.
5.
Set the timeout (40 seconds by default). When the timeout expires, the call falls back to the original extension.
6.
Set the number of digits for parking positions.
NOTE
se the same number of digits as for extensions (your dial plan). Users can park calls on positions which match their extension numbers.
7.
Save your settings.
Using call parking
1.
Initiate or answer the call.
2.
Select the call transfer function on your phone.
For more information, refer to Hardware telephone basic usage
3.
Dial the call parking number. You can:
Dial the whole parking slot number (for example,
*512
) to park the call to the specific slot.
Dial the
Call parking
extension only (for example,
*5
) to park the call in the first available parking slot. The voice-prompt message tells you the number of the first available parking slot.
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4.
Select the call transfer function on your phone.
5.
Terminate the call.
To answer a parked call:
1.
Pick up the phone.
2.
Dial the call parking number (for example,
*512
).
If nobody answers the parked call before the timeout expires, the call falls back to the original extension.
4.5.5 Configuring and using conferences
Telephone conference is one telephone call of three or more users.
Telephone conferences allow participation of Kerio Operator users and external participants. To join a conference, participants must dial the conference number and PIN.
You can use two different types of conferences — statically or dynamically configured.
Statically configured conference
Statically configured means that conferences are created in the administration interface and each new conference uses one extension.
NOTE
If there is a lack of extensions, use dynamically configured conferences instead.
Configuring statically configured conferences
1.
Go to section
Status > Dial Plan
and make sure that the extension you have selected for the conference is not used.
2.
In
Configuration > Conferences
, click
Add
. The
Add conference
dialog is dispayed.
3.
Enter the conference extension and its description.
4.
In the menu
Conference type
, choose the
Statically configured
option.
5.
Optional: Limit the number of participants.
6.
Each conference can be protected by a PIN required from all participants upon attempting to enter the conference. If you wish to secure a conference, set a PIN and deliver it to the members.
7.
To enable call recording, select
Record Calls
.
WARNING
Please note that call recording is a subject to special laws in many countries. It maybe illegal in your jurisdiction or require notice to the other party on the call. Accordingly, you assume all liability for using the call recording functions and are responsible for notifying all users of this system of this potential restriction, if applicable.
Connecting to a statically configured conference
1.
Dial the conference telephone number / extension.
2.
If the conference is protected, you will be asked to enter the PIN.
To leave the conference, simply terminate the call.
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Dynamic conferences
A dynamic conference is created on one extension only. Users set the conference number and PIN after dialing the extension or the whole telephone number. On one extension, users can set unlimited number of conferences with different conference numbers.
The disadvantage of dynamic conference is that user has to enter three numbers when dialing the conference (the extension, the conference number and the PIN).
Configuring dynamic conferences
1.
Go to
Status > Dial Plan
and make sure that the conference extension is not used by a user.
2.
In
Configuration > Conferences
, click
Add
. The
Add conference
dialog is dispayed.
3.
Enter the conference extension and its description.
4.
In the
Conference type
menu, choose option
Dynamic, created on demand
.
5.
To enable call recording, select
Record Calls
.
WARNING
Please note that call recording is a subject to special laws in many countries. It maybe illegal in your jurisdiction or require notice to the other party on the call. Accordingly, you assume all liability for using the call recording functions and are responsible for notifying all users of this system of this potential restriction, if applicable.
Connecting to a dynamic conference
To connect to an existing conference, enter the conference number and PIN (if required).
Creating a dynamic conference
1.
Dial the conference telephone number / extension.
2.
Enter any number for the conference.
3.
Set PIN (if required).
4.
Communicate these access numbers (extension, conference number and PIN) to other attendees.
To leave the conference, simply terminate the call.
Where to monitor conference activities
All current calls can be viewed under
Status > Conferences
.
4.5.6 Configuring auto attendant scripts
Auto attendant script is a simple collection of voice menus, submenus and announcements and actions defined for each of them according to the caller's behavior. It can: connect to an extension or voicemail, play an announcement, navigate through menus and submenus.
send any faxes to a configured email.
Menus can be recorded in various formats. Kerio Operator supports the following formats:
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Supported formats
gsm wav
Audio format
8KHz
8KHz, 16 bits per sample, mono (Kerio Operator encodes all WAV files into this format automatically)
Screenshot 46: Kerio Operator — supported audio formats
How to add new auto attendant script
See the following description of an auto attendant script as an example. Create a script which: starts after dialing extension 200, contains a voice menu with the following text: LOL! You have just reached the Live And Let Laugh company's hotline
(fiendish laugh).
For Sales Department, press 1.
For Quality Assurance Department, press 2.
For Technical Support Department, press 3.
If you wish to speak to the receptionist, press 4.
The Sales Department manages two flagship products of the company. Therefore, two submenus (
Joke Lite
,
Laugh
Home 2012
) are created.
For
Joke Lite
, press 1.
For
Laugh Home 2012
, press 2.
If you wish to talk to the receptionist, press 3.
Create the same menu for technical support.
Before creating the script, it is necessary to create extensions (in the assigned range 123456XXX) which will be used in the script.
extension 100
— reception of Live And Let Laugh Inc. One of the receptionists Joan Giggle or Brian Snigger will connect the calls if the caller makes no selection from the menu.
extension 203
— Quality Assurance Department extension (David Jester).
extension 301
— common extension (you can create a call queue or a ringing group) for Joke Lite experts, such as
Frederic Jovial, George Funpoker, Anne Kdotte.
extension 302
— common extension for Laugh Home 2012 experts (Tamara Bellylaugh, Otto Spass, Mary Merry).
extension 501
— call queue for Joke Lite technical support (Andrew Widegrin).
extension 502
— call queue for Technical Support of Laugh Home 2012 (Alan Tickle).
Script settings
Configure the script in the administration interface in section
Configuration > Auto Attendant Scripts
:
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Screenshot 47: Auto Attendant
1.
Click
Add
and enter the
Script extension
(extension
200 in our example) and some description.
2.
(Optional) To receive faxes to configured email address, select
Detect faxes and send them by email to
and type an email address.
3.
Click
Edit
and open the
Edit Menu
dialog.
4.
In the
Announcement
field, select the recording for the main script. The
Select
button offers existing recordings or you can upload your own announcement to the PBX.
NOTE
If you open the administration interface in Safari browser and you cannot play any recordings, read topic
Cannot play voicemails or audio files in Safari
5.
Set
Number of playbacks
to two which will ensure the menu is played to the caller twice.
6.
Once the announcement is played, timeout is started with the default action taken upon its expiration. Set the timeout to 10 seconds. The default action is the preset hang up action. This means that if the announcement is played twice and the customer does not make any selection within 10 seconds, the call will be terminated.
7.
Click
Add
. The
Key
column states the key which confirms the customer's choice. Enter number 1. Enter 1 in this column. Column
Action
defines what happens when the caller presses a key on their phone. Select
Go to submenu
.
We need to direct calling customers to the extension of the product they are interested in (either Joke Lite or Laugh
Home 2012). In the
Announcement
column, you can add a record which will be played upon pressing the particular key (for example: Stay tuned, now you will be redirected to the Live And Let Laugh Inc Sales Department). Finish the table according to figure.
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Screenshot 48: Editing main menu
8.
Check
Interpret any other input as extension number and dial it
. This option allows to specify a direct extension while the auto attendant script is running.
9.
Confirm the settings and return to the
Add Auto Attendant Script
dialog which is now similar to the one in picture above.
10.
Click on menu
Sales dept.
. Again, the
Edit menu
dialog is opened but now the menu is for the Sales department.
Follow the same procedure as with the main menu. The resultant menu will look as the one showed in the picture below.
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Screenshot 49: Submenu edit
11.
Do the same for the
Technical Support dept.
menu.
Now the script is complete.
NOTE
You can duplicate an existing script if you want to create a similar one — select a script and click
Duplicate
.
Time condition
The script can be limited to a specific time interval (office hours of your employees or night time when no call queue agents are available).
are configured in section
Configuration > Definitions > Time Ranges
. Once you have the time range configured, go back to the
Add Auto Attendant Script
, select the menu you wish to limit and click on the
Convert to Time Condition
button.
Instructions for time condition setting will be better understood through the following example focusing company's working hours. Sales department works from 9am to 5pm on weekdays. Configure the auto attendant script so that when customers call during office hours they will be connected to a sales department employee and when they call before or later they will hear a message announcing that the sales department is closed. To create the condition script, follow these instructions:
1.
In the administration interface, go to
Configuration > Definitions > Time Ranges
.
2.
Click
Add
. Dialog
Add Time Range
opens.
3.
In section
Add to a group
, select the
Create new
option and enter a name for the new interval (for example,
Sales Department Office Hours
).
4.
The
Description
is optional, for example
Weekdays from 9am to 5pm
.
5.
Select daily in the
Type
menu and set the desired interval from 9 to 5 in the
From
and
To
fields.
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6.
In the
Valid on
menu, select
Weekdays
.
7.
Click
OK
to confirm the changes.
8.
Open the
Configuration > Auto Attendant Scripts
section.
9.
Click on
Add
.
10.
In the
Add Auto Attendant Script
dialog, create a corresponding menu (the script created in the previous section will be used in this example — see the picture below).
11.
Select the
Sales Department
submenu and click
Convert to Time Condition
.
12.
Divide the Sales Department submenu in two time conditions. The first one is played if the condition is met and the second if the condition is not met. Click on the red highlighted text
Set up the time condition
.
Screenshot 50: Setting the time condition
13.
This opens dialog
Edit Time Condition
. In the
For time range
menu, select
Sales Department Office Hours
.
14.
Click on the submenu representing the positive result of the condition. It is currently called
Unnamed
. In the dialog
Edit Menu
just opened, simply add a description (for example
Sales Department condition met
).
15.
Click on the submenu representing the negative part of the condition (now it is empty and unnamed).
16.
This opens dialog
Edit Time Condition
allowing to add a description (for example, Sales Department — condition not met).
17.
Now you can modify the script. For example, in the
Announcement
field, add a message announcing that office hours of the Sales Department are from 9am to 5pm on weekdays.
18.
Save the submenu. The resultant script is displayed in the next picture.
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Screenshot 51: Time condition applied in the script
4.5.7 Setting time conditions in auto attendant scripts
Time conditions are best explained in an example
When configuring auto attendant scripts, Bob encountered the following problem. The company management created a new quality department. The responsible person is Alice. Bob created a new extension for this department. Alice came to Bob complaining that dissatisfied customers are calling constantly and she does not even have time for lunch.
Bob knew that Alice needs an auto attendant script which respects her working hours. And how to do it?
1.
Bob created
for Alice’s working hours, her lunch break and also for public holidays.
2.
He
for the following announcements:
Hello. You are calling Live And Let Laugh Inc. We are having a delicious lunch at the moment. If you call after
1pm, we will gladly hear what you have to say. Talk to ya later!”
“Hello. You are calling Live And Let Laugh Inc. We are off the clock at the moment. Please, call us on weekdays from 8am to 12pm or after lunch from 1pm to 6pm. We will gladly hear what you have to say. Talk to ya later!”
“Hello. You are calling Live And Let Laugh Inc. Have a very merry holiday today. If you wish to make a complaint, call us on weekdays from 8am to 12pm or after lunch from 1pm to 6pm. We will gladly hear what you have to say. Talk to ya later!”
3.
He created a new auto attendant script with time conditions
NOTE
You can also use the Day/night mode to create time conditions without a specific time set.
For more information, refer to Using the Day/night mode in auto attendant scripts (page 241).
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Setting time intervals for auto attendant scripts
1.
In the administration interface, go to section
Definitions > Time ranges
.
2.
Add three new time ranges. Two ranges will be of the daily time —
Lunch break and
Working Hours
. Both ranges will be valid on weekdays.
3.
The third range will be absolute. Add the first public holiday when creating the range. Add also other public holidays and do not forget to add them into the existing group
Holidays
.
Creating auto attendant scripts in Kerio Operator
The script will follow this scheme:
If Holidays
Else publicholidays.wav
If Working hours
If Lunch break lunchbreak.wav
Else
Action: Redirecting to Alice’s extension.
Else offtheclock.wav
1.
In the administration interface, go to
Auto Attendant Scripts
.
2.
Add a new script, assign it extension 300 and add a description (
Scripts for complaints desk
).
3.
Create the first condition: Click
Convert to Time Condition
. Double-click on the red link
Set up the time condition
and in the
Edit Time Condition
dialog, select range
Holidays
. Save the settings.
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4.
Now, edit the first part of the condition in dialog
Add Auto Attendant Script
. Double-click .
5.
In the
Edit Menu
dialog, type description
Holidays and add a file with the announcement about a holiday. Set timeout to 5 second (this will suffice) and save the settings.
6.
Create the second condition: Select the icon and click
Convert to Time Condition
(thus the Working hours condition will be nested into condition Holidays). In the
Description
field, enter
Working hours
; in the
For time range
, select
Working hours
. Save the settings.
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7.
In the
Edit Auto Attendant Script
dialog under the
Working hours
line, two new conditions appear.
8.
Create the third condition: Click select
Lunch break
. Save the settings.
and click
Convert to Time Condition
. In the
For time range
menu,
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9.
Double-click the last icon. In the
Edit Menu
dialog, type description
Lunch break
and add a file with the announcement about a lunch break. Set timeout to 5 second and save the settings.
10.
Double-click the icon (last but one in the scheme). In the
Edit Menu
dialog, type description
Working hours (dial Alice)
. You can add an
Announcement
with information about redirecting to the Complaints department.
Set
Timeout
to 1 second. In the
Default action
menu, select
Dial extension number
. Type Alice’s extension (211) in the
Extension
field and save the settings.
11.
Double-click the last condition (icon ). In the
Edit Menu
dialog, type description
After Hours
and add a file with the announcement that the Complaints department is close at the moment. Set timeout to 5 second and save the settings.
WARNING
If you open the administration interface in Safari browser and you cannot play any recordings, read topic
Cannot play voicemails or audio files in Safari .
The resultant script is displayed below.
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4.5.8 Using the Day/night mode in auto attendant scripts
NOTE
New in Kerio Operator 2.3.2!
Day/night mode works similarly to time ranges in
. The difference is that in a time range you set a specific time. The Day/night mode is switched on demand.
An example:
Alice works in a Sales department and has flexible working hours. She needs her extension to be available only when she's at work. She cannot use
because her working hours are not fixed. She will use the Day/night mode to switch her extension on and off easily.
Alice must
add a Day/night mode service to Kerio Operator
and then create an
auto attendant script using the created service .
Adding a Day/night mode
1.
In the administration interface, go to section
Configuration > PBX Services
.
2.
Click the
Add > Add a Day/night Mode Service
.
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3.
In the
Add Service
dialog window, fill in the service extension and type a name.
4.
Click the two keys icon to display the service PIN number.
5.
Click
OK
.
You can create as many Day/night mode services as you need, for example one for each department.
Selecting the Day/night mode status
You can select the mode status either manually in the administration interface or by calling the service extension.
The
Configuration > PBX Services
section displays the current status of each mode in the service list.
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Selecting the status in the administration interface
1.
In the administration interface, go to section
Configuration > PBX Services
.
2.
Double-click the desired Day/night mode service.
3.
Change the mode status.
4.
Click
OK
.
Selecting the status by calling the extension
1.
Call the service extension (
88 in our example).
2.
Enter the service PIN code. When you switch to the day status, you hear a ascending sound. When you switch to the night status, you hear a descending sound.
3.
Hang up.
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Using day/night mode in auto attendant scripts
The script will follow this scheme:
If Working hours
Else
Redirect to extension 256
Play offtheclock.wav
Hangup
1.
In the administration interface, go to
Auto Attendant Scripts
.
2.
Add a new script, assign it extension 400 and add a description (
Working hours Sales dept.
).
3.
Click the
Convert to Condition
button. The
Edit Condition
dialog opens.
4.
Select the
Day/night mode
mode from the drop-down menu. Click
OK
.
5.
Select the Day status (the sun icon) and click
Edit
. Set
Dial extension number
as the
Default action
. Type the extension number and click
OK
.
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6.
Select the Night status (the moon icon) and click
Edit
. Select the
Announcement
to be played when the night status is active ( offtheclock.wav
in our example). Set
Hang up
as the
Default action
. Click
OK
.
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WARNING
If you open the administration interface in Safari browser and you cannot play any recordings, read topic
Cannot play voicemails or audio files in Safari .
The resultant script is displayed below.
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Now Alice calls the Day/night mode extension (
88 in our example) when she comes to work to switch the day status on. She calls the extension again to switch to the night status when she goes home.
4.5.9 Configuring call pickup
This function enables users to answer a call ringing on an extension on a device at another extension. The PBX distinguishes between two types of call pickup:
Call pickup within defined groups (so called rooms) by using specific code (by default, this code is
*8
),
Call pickup by using a special code (by default, this code is **) with the called extension appended at the end.
How to configure call pickup rooms
1.
In the administration interface, go to
Configuration > PBX Services
, enable
Call Pickup
and save the settings. Keep the default pickup code (
*8
) unless you do have a reason to change it.
2.
Go to
Definitions > Call Pickup Rooms
and click
Add
to open dialog
Add Call Pickup Room
.
3.
Type the name of the department or the office in the
Name
field.
4.
In the table, add all users and extensions that will be able to pick up calls for one another.
5.
Make sure the
Room is enabled
option is checked.
Example
The Live And Let Laugh company network administrator uses the
Add Call Pickup Room
dialog to add a group with room name Local Sales for HPR (Happy people Republic). He adds all sales assistants for local market and their extensions: Frederic Jovial, Mary Merry, George Funpoker.
Frederic Jovial has a day off today. His phone is ringing. Thanks to the call pickup rooms feature, Mary Merry does not need to dash for the Frederic’s desk every time a customer calls his extension. She simply dials the magic code
*8 and serves the customer at her desk.
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How to configure directed call pickup
Directed call pickup
is a service allowing to pickup calls directed to any extension at the PBX. Imagine the following situation: the managing director Peter Prank uses extension
101 the financial director Oscar Jape uses extension
102 they share an assistant, Ms Alessandra G. Uffaw.
If Alessandra's phone shows information that someone is calling the managing director (Peter Prank) during his meeting with the financial director (Oscar Jape), she can accept the call by dialing
**101
. Once she picks up the call, she learns that the caller is the International laughter Association manager and arranges a meeting for him and her company’s executive manager. A few minutes later, the phone at the desk of the financial director Oscar Jape starts ringing. Again, the assistant can accept this call at her desk phone. now she enters the code
**102 and recommends the caller (the
Cirque de Rire ringmaster) to call Mr Jape back later.
As you can see, by dialing the call pickup code, you can answer a call for any extension of the PBX.
For directed call pickup, apply settings as described below:
1.
In the administration interface, go to
Configuration > PBX Services
.
2.
Enable
Directed Call Pickup
.
3.
Directed call pickup is now fully functional.
NOTE
You can use directed call pickup in Kerio Phone .
4.5.10 Configuring call queues
Call queues are used to distribute incoming calls between agents.
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Screenshot 52: Call queue
Configuring call queues
1.
In the administration interface, go to
Configuration > Call Queues
.
2.
Click
Add
to open the
Add Call Queue
dialog. On the
General
tab, type the new queue extension number.
3.
Select the
.
4.
Click the
Agents
tab.
5.
If you want your agents to log in
dynamically , type login and logout code. For example, 12345 to login, and 54321 to
logout. The calls will only go to agents logged into the queue. If you want to assign specific agents permanently to the queue, click
Add
to select their extensions.
NOTE
Both methods can be combined. One queue may have agents who are assigned permanently and agents who log in dynamically.
6.
Click the
Announcements
tab. An announcement is a pre-recorded message that callers hear while waiting in a call queue. You can import pre-recorded announcements into Kerio Operator (see topic
Language settings in Kerio Operator
Configuration > PBX Services > Record audio
(see topic
).
How to select a queue strategy
Round robin with memory mode uses circular call distribution. It remembers the last agent who answered the phone, and new calls are directed to the next agent in the round queue.
Ring all agents — calls always ring at all agents until one of them answers the particular call.
Ring least recently called agent — the system selects the agents who have not answered the phone for the longest period.
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far.
Ring agent with fewest calls — the system assigns the call to the agent with the lowest number of calls answered so
Ring random agent — if you select this option, the system will choose an agent randomly.
Ring in order — only for permanently assigned agents. You specify a fixed order in which they are always selected.
This strategy is for companies where all calls are answered by a receptionist. In case the receptionist is not answering, the call is directed to the next agent in order (for example, an administration assistant).
What is the difference between permanently assigned and dynamic agents
Permanent assignment — agent's extension is assigned permanently to the queue.
Dynamic login — agents use special code for logging in and out of the queue.
Recording calls from call queues
Kerio Operator allows recording calls from call queues. No other module or equipment is necessary. Setting can be done as follows:
1.
Open the
Configuration > Call Queues
section and select the queue in which you wish to record the calls.
2.
On the
General
tab, select
Record calls
.
To play back recorded calls
WARNING
Please note that call recording is subject to special laws in many countries. It may not be legal in your jurisdiction, or may require notice to the other party on the call. Accordingly, you assume all liability for using the call recording functions and are responsible for notifying all users of this system of this potential restriction, if applicable.
Section
Status > Recorded Calls
displays all calls recorded from call queues. Select a call to listen to it, download it to your computer or remove it.
Deleting Recorded Calls
Recorded calls can be periodically deleted once their total size reaches a certain limit. The limit can be set in section
Status > Recorded Calls
.
1.
Click
Advanced > Periodically Remove Old Recorded Calls
.
2.
In
Remove Old Recorded Calls
dialog box, enter the maximum size of recorded calls on a disk (in MB).
Configuring a call queue timeout
The call queue timeout period determines the maximum amount of time a caller can be placed in a call queue.
Configuring the limit prevents from waiting in a queue infinitely.
The timeout limit is unlimited by default. For setting the limit, perform these steps:
1.
In administration, go to
Configuration > Call Queues
.
2.
Click
Add
/
Edit
.
3.
On tab
General
, set
Queue timeout
.
4.
(Optional) Go to tab
Announcements
and select
Timeout announcement
. Such announcement will play when the limit is reached and should include information about what happens next (tab Exceptions).
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5.
Go to tab
Exceptions
.
6.
Choose an action for exceeded limit:
Callers receive a busy signal
— if announcement was set, recording plays before call termination.
Forward to
— type an extension. Kerio Operator forwards callers to the extension. If announcement was set, recording plays.
7.
Save the settings.
Timeout is configured. If you want to check your settings, lower the limit to several seconds and dial the queue from several phones.
Configuring a music on hold and a while waiting period
A while waiting period is the period when users are waiting in a call queue for an agent. You can set what is playing during the period:
1.
In administration, go to
Configuration > Call Queues
.
2.
Click
Add
/
Edit
.
3.
On tab
General
, select
While waiting
:
Music on hold — a music sounds during the while waiting period.
Ringtone — a ringtone sounds during the while waiting period.
4.
If you selected the
Music on hold
option, select the particular recording in the
Music on hold
menu. If you want to add a new recording to Kerio Operator, go to the
Definitions > Music on Hold
section.
5.
Save the settings.
Configuring a queue length
A queue length determines max. number of callers in the queue at the same time. Configuring the limit prevents from waiting too long in the queue.
The queue length is unlimited by default. For setting the limit, perform these steps:
1.
In administration, go to
Configuration > Call Queues
.
2.
Click
Add
/
Edit
.
3.
On tab
General
, set
Queue length
.
4.
(Optional) Go to tab
Announcements
, select
Full queue announcement
. Such announcement will play when the limit is reached and should include information about what happens next (tab Exceptions).
5.
Click the
Exceptions
tab.
6.
Select an action for exceeded limit:
Callers receive a busy signal
— if an announcement was set, recording plays before a call is terminated.
Forward to
— type an extension. Kerio Operator forwards callers to the extension. If an announcement is set,
Kerio Operator plays the recording.
7.
Save the settings.
The queue length is configured. If you want to check your settings, lower the limit to 1 and dial the queue from two phones.
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Configuring exit keys
You can set exit keys for each call queue. Callers can use an exit key for transfer to an extension.
1.
In administration, go to
Configuration > Call Queues
.
2.
Click
Add
/
Edit
.
3.
On tab
General
, click
Edit
next to
Exit keys
.
4.
Edit
Exit Keys
, click
Add
.
5.
In the
Add Exit Key
dialog, type an exit key (for example 1).
6.
Type an existing extension to transfer calls.
7.
Type a description.
8.
Save the settings.
When users standing in the queue use the exit key, they are transferred to pre-configured extension.
Configuring call queues without agents
Follow these steps if no agents are logged into the queue:
1.
In administration, go to
Configuration > Call Queues
.
2.
Click
Add
/
Edit
.
3.
(Optional) Go to the
Announcements
tab, select
No agents announcement
. Kerio Operator plays the announcement when there are no agents in the queue.
4.
Go to tab
Exceptions
.
5.
Select an action if the queue has no agents:
Callers cannot join the queue. Callers already waiting are removed
— Kerio Operator disconnects all callers. If
No agents announcement
is selected, Kerio Operator plays the recording.
Callers can join the queue
— new callers can connect to the queue. Current callers stay in the queue. If
No agents announcement
is selected, Kerio Operator plays the recording.
Callers cannot join the queue
— new callers cannot connect to the queue. Current callers stay in the queue. If
No agents announcement
is selected, Kerio Operator plays the recording.
6.
If you selected
Callers cannot join the queue
or
Callers cannot join the queue. Callers already waiting are removed
, select one of these actions:
Callers receive a busy signal
Forward to
— type an extension or external phone number. Kerio Operator forwards callers to the number.
7.
Save the settings.
Settings are complete now. If you want to check your configuration, test these cases:
1.
No agent serves the queue. Try to join the queue as a caller.
2.
One agent serves the queue. Join the queue as a caller. Agent logs out.
Prioritizing call queues
Agents can operate several call queues. In the following example, an agent is assigned to three queues.
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Screenshot 53: Operating multiple queues at once
To help agents identify the queues, you can upload various audio records for each queue. The record identifying the queue is played to the agent before a call from this queue is connected.
Upload new audio record as follows:
1.
Select a call queue or create a new one in section
Configuration > Call Queues
.
2.
In the displayed dialog, go to the
Announcements
tab.
3.
Check the
Help agents identify the source queue by playing this announcement
and click on
Select
.
4.
In the
Select Audio File
dialog box, double-click a record to select it, or upload your own record to Kerio Operator (it must be in WAV or GSM format). Use the
Upload
button.
It is also possible to set priorities for individual queues:
1.
Open the
Configuration > Call Queues
section.
2.
Select a queue or create a new one.
3.
In the displayed dialog, go to the
General
tab and set the desired priority.
4.
Repeat the configuration for other queues.
Queues with higher priority are processed first.
Displaying missed calls on phones in call queues
When an incoming calls rings in the call queue and an agent answers it, other devices in the queue register the call as missed anyway.
To not display missed calls on other devices:
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1.
In the Kerio Operator administration interface, go to
Configuration > Call Queues
.
2.
Select an extension and click
Edit
.
3.
Switch to
Advanced
tab.
4.
Select
Do not display missed calls on the phones
.
5.
Click
OK
.
Monitoring active call queues
1.
In the administration interface, go to section
Status > Call Queues
.
2.
The top table shows currently active queues.
3.
The other tables display agents and callers in a queue. Just select a queue and the details in table
Agents
and
Callers
are updated.
You can also reset the call queue statistics to start from zero. Use the
Reset Statistics
button.
4.6 Security
This section helps you secure your Kerio Operator server.
4.6.2 Configuring SSL certificates
4.6.1 Securing Kerio Operator
Issues to address
Restrict communication on the integrated firewall in Kerio Operator.
Restrict the number of attempts to enter SIP passwords.
Using special rules, forbid international outgoing calls to countries you do not communicate with
Restrict international outgoing calls to countries where you rarely call
The following sections describe these settings in detail.
Configuring firewall in local network
Kerio Operator is usually protected by firewall (in your local network or in the Internet). Certain ports need to be opened
(or mapped) on firewall.
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Service (default port)
SIP (5060)
IMAP (143)
LDAP (389)
LDAPS (636)
HTTP (80)
HTTPS (443)
HTTPS (4021)
Outbound connection
allow allow if integration with Kerio Connect is enabled and there is a firewall between Kerio Connect and Kerio Operator.
allow allow allow allow allow allow
Inbound connection
allow for SIP servers of your provider deny deny allow if you use mapping from Active Directory or Open Directory and there is a firewall between the directory service and Kerio Operator.
deny allow if you wish users to be able to connect to
Kerio Phone from the Internet.
allow if you wish users to be able to connect to the administration interface from the Internet.
allow STUN/TURN
(3478)
STUN/TURN
(3479) allow allow
Configuring firewall integrated in Kerio Operator
Prepare groups of IP addresses which you wish to allow for individual services (create them in
Definitions > IP Address
Groups
).
You can configure the integrated firewall in section
Network > Firewall
.
Service
Web server
SIP
Phone provisioning
CRM integration
SNMP monitoring
Recommendation
If you want to restrict connections to Kerio Operator administration and softphone, check this option and select an IP group with addresses from which access will be allowed. Bear in mind that all the PBX users should be allowed to connect to Kerio Phone at least from their own workstation.
We recommend to restrict the SIP protocol solely to your internal network and external IP addresses of your SIP provider.
For security reasons, we recommend to restrict automatic phone provisioning solely to your internal network because TFTP sends configuration data as plain text.
For security reasons, we recommend to restrict communication solely to your internal network.
For security reasons, we recommend to restrict communication solely to your internal network and IP adressess where monitoring servers are running.
NOTE
If the options are unchecked, no restrictions are set.
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Configuring protection against password guessing
Login data guessing is one of the most common attacks on a PBX. In Kerio Operator, attackers try to guess extension numbers and SIP passwords. This type of attack is defined by many unsuccessful attempts to enter extension number and SIP password during a login. Kerio Operator security settings enable you to limit the number of attempts of a phone
(both software and hardware) to connect to the PBX. Apply settings as described below:
1.
In the administration interface, go to
Security
.
2.
Set the limit of unsuccessful attempts (usually 3 to 10 attempts) and set the time period during which attempts will be counted. Setting the time period protects real users who have forgotten their password or who have made mistakes during several logins. When the time limit expires, they can try to login to the PBX again.
3.
Set the time during which Kerio Operator will block the source IP address.
4.
You can also enter an email address that will be used for sending warnings about blocked IP addresses.
How to recognize there has been an attack attempt
In log
Security
look for the
Authentication failed string. If there are many messages of this kind, somebody is trying to use a dictionary attack.
What to do in case of an attack
In case of an attack, apply the following instructions as soon as possible:
1.
In section
Status > Calls
and in logs, look for information on which account has been abused.
2.
Change the SIP password of this account.
3.
Instruct users about handling their login details and secure behavior on the Internet.
4.
The PBX is blocked, so it needs to be unlocked again.
4.6.2 Configuring SSL certificates
To secure the PBX by SSL/TLS encryption, you need a SSL certificate. SSL certificates authenticate an identity on a server.
Kerio Operator creates the first self-signed certificate during the installation. The server can use this certificate but users will have to confirm they want to go to an untrustworthy page. To avoid this, generate a new certificate request in Kerio
Operator and send it to a certification authority for authentication.
WARNING
If you use the Safari browser in your environment (on Apple iPhone, Apple iPad, Mac OS X systems and on Microsoft
Windows), you will not be able to play voice messages in Kerio Phone on their devices with a self-signed certificate.
You must have a trustworthy certificate available.
If you use a self-signed certificate, users with Apple mobile devices will not be able to play voice messages in Kerio
Phone on their devices. They must have a trustworthy certificate available.
To encrypt the communication between Kerio Operator and hardware phones (and only a self-signed certificate available), you have to import or configure information in the phones that the invalid certificate is to be ignored.
Creating self-signed certificates
To create a self-signed certificate, follow these instructions:
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1.
In the Kerio Operator administration interface, open section
Definitions > SSL Certificates
.
2.
Click
New > New Certificate
.
3.
In the
New Certificate
dialog box, type the hostname of Kerio Operator, the official name of your company, city and country where your company resides and the period of validity. The
Hostname
and
Country
entries are required fields.
4.
Click
OK
.
5.
To enable the server to use this certificate, select the certificate and click
Set as Active
.
Creating certificates signed by certification authority
If you wish to create and use a certificate signed by a trustworthy certification authority, follow these instructions:
1.
In the Kerio Operator administration interface, open section
Definitions > SSL Certificates
.
2.
Click
New > New Certificate Request
.
3.
In the
New Certificate Request
dialog box, type the hostname of Kerio Operator, the official name of your company, city and country where your company resides and the period of validity. The
Hostname
and
Country
entries are required fields.
4.
Click
OK
.
5.
Select the certificate and click
Export
.
6.
Save the certificate to your disk and email it to a certification organization.
7.
Once you obtain your certificate signed by a certification authority, go to
Definitions > SSL Certificates
.
8.
Click
Import
.
9.
To enable the server to use this certificate, select the certificate and click
Set as Active
.
Intermediate certificates
Kerio Operator supports
intermediate
certificates.
To add an intermediate certificate to Kerio Operator, follow these steps:
1.
In a text editor, open the server certificate and the intermediate certificate.
2.
Copy the intermediate certificate into the server certificate file and save. The file may look like this:
-----BEGIN CERTIFICATE-----
MIIDOjCCAqOgAwIBAgIDPmR/MA0GCSqGSIb3DQEBBAUAMFMxCzAJBgNVBAYTAl
MSUwIwYDVQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMR0wGwYDVQ
..... this is a server SSL certificate ...
ukrkDt4cgQxE6JSEprDiP+nShuh9uk4aUCKMg/g3VgEMulkROzFl6zinDg5grz
QspOQTEYoqrc3H4Bwt8=
-----END CERTIFICATE-----
-----BEGIN CERTIFICATE-----
MIIDMzCCApygAwIBAgIEMAAAATANBgkqhkiG9w0BAQUFADCBxDELMAkGA1UEBh
WkExFTATBgNVBAgTDFdlc3Rlcm4gQ2FwZTESMBAGA1UEBxMJQ2FwZSBUb3duMR
..... this is an intermediate SSL certificate which signed the server certificate...
5BjLqgQRk82bFi1uoG9bNm+E6o3tiUEDywrgrVX60CjbW1+y0CdMaq7dlpszRB
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t14EmBxKYw==
-----END CERTIFICATE-----
3.
In the administration interface, go to section
Definitions > SSL Certificates
.
4.
Import the modified server certificate by clicking on
Import > Import a New Certificate
.
NOTE
If you have multiple intermediate certificates, add them one by one to the server certificate file.
Securing Kerio Phone with SSL certificates
For more information about securing Kerio Phone, see the
Securing Kerio Phone with SSL certificates
section in the
Provisioning for Kerio Phone
topic.
4.6.3 Configuring NAT
Kerio Operator is behind NAT and phones are in the Internet
1.
In the administration interface, open section
Network > General
.
2.
In the NAT support section, enable NAT by checking the option.
3.
Enter the public address which should be used in SIP protocol messages.
4.
For phones in the same private network as Kerio Operator, create an appropriate IP address group in section
Configuration > Definitions > IP Address Groups
with all addresses on which phones communicate in your private network. Thus, the PBX will communicate with phones within the network directly.
5.
(Optional) You can also limit the RTP port range. Each call requires 4 ports for communication.
6.
Also, map the following ports from firewall to Kerio Operator. It is usually necessary to map a port range for RTP (according to the specified interval).
TCP+UDP/5060
TCP/5061
UDP/443
TCP+UDP/3478
TCP+UDP/3479
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Screenshot 54: Kerio Operator is behind NAT and hardware phones are in the Internet
Kerio Operator is in the company network and hardware phones are behind NAT
Firstly, configure
The scenario in figure bellow requires only one minor configuration in the PBX settings:
1.
In the administration interface, open the
Extensions
section.
2.
Select the extension of the user whose phone is in a private network.
3.
In the
Edit extension
dialog, go to tab
Advanced
.
4.
Check the
Extension is behind NAT
option.
Screenshot 55: Kerio Operator is in the company network and hardware phones are behind NAT
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Kerio Operator is behind NAT and hardware phones are in the Internet
Firstly, configure
If the telephone is in the Internet (not behind NAT), Kerio Operator does not require special configuration.
WARNING
Phones which are in the Internet cannot be managed in section
.
4.7 Server settings
This section contains information about:
4.7.1 Language settings in Kerio Operator
4.7.2 Configuring Built-in DHCP server in Kerio Operator
4.7.3 Configuring parameter 66 in DHCP server in Kerio Control
4.7.4 Configuring server date, time and time zone in Kerio Operator
4.7.5 Configuring standard phone interfaces
4.7.6 Connecting Kerio Operator to directory service
4.7.7 Connecting multiple Kerio Operators
4.7.8 Routing calls between multiple Kerio Operators and the PSTN
4.7.9 Creating and using speed dial
4.7.10 Creating ringing groups
4.7.11 Customization of voice sets
4.7.12 Customizing the Kerio Phone login page
4.7.13 Distinctive ringing support
4.7.14 Fax support in Kerio Operator
4.7.16 Integrating Kerio Connect and Kerio Operator
4.7.17 Setting optional call recording
4.7.18 Setting outgoing calls constraints in Kerio Operator
4.7.20 Using paging groups and services
4.7.21 Configuring Click to Call in Kerio Connect client
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4.7.1 Language settings in Kerio Operator
Languages in Kerio Operator are:
Application language — language for the administration interface and for Kerio Phone.
PBX language — the voice of the PBX. Voice records which are used for communication with users (internal and external).
You can also change the type of indication tones according to individual countries (read section
Changing indication tones according to countries ).
Changing the application language
The language for the administration and softphone interfaces can be set in the
Admin
menu in the right top corner of the of the application window.
Changing the language of the PBX
You can change the default language of the PBX in the administration interface in section
Configuration > Advanced
Options > Telephony
.
There, you can also upload
or different voice records of the same language (for example, less formal records).
When setting language, bear in mind the following rules:
Default language set in section
Advanced Options > Telephony
has lower priority than settings of individual users in section
Users
. If users do not have any language set, the default one is used.
Default language set in section
Advanced Options > Telephony
has lower priority than settings for interfaces for incoming calls (section
Call Routing
). The language set for the interface of incoming calls has lower priority than files uploaded to call queues (see screenshot below). If no language is set, the default one is used. The same goes for call queues.
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How to change the language for individual users
Thomas Punchline, the network administrator at Live And Let Laugh Inc, faces the following problem: New employee has arrived in the company. Alessandra G. Uffaw has moved from the Bliss Seekers Land to the Happy People Republic and cannot speak the Happish language. She complains she can’t understand her voicemail. Thomas has to switch the
PBX language to the Cravish language for her. Do you need to solve a similar problem? Check the following example:
1.
In the administration interface, go to
Configuration > Users
.
2.
In the user’s settings, go to tab
General
and change the
Phone language
.
How to change the PBX language for a group of users
Thomas was instructed to create a new interface in Kerio Operator and change its language to the Cravish. He has to create a new interface for incoming calls and set a language for this interface. He called his VoIP service provider and purchased new phone numbers for the employees who will communicate with foreign customers. And how he configured Kerio Operator?
1.
In the administration interface in section
Configuration > Call Routing
, add a new route for incoming calls.
2.
Connect it to the provider, open the edit dialog by clicking on the route in table
Interfaces and routing of incoming calls
.
3.
Select a language on tab
Advanced
.
4.
Select a country on tab
Advanced
. Each country has different standards for indication tones during calls (e.g. beeps, ringing tones, etc.).
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Setting a different language for a call queue
If you wish to change the language for call queues, not for the entire route, go to section
Configuration > Call Queues
.
NOTE
Language files used in call queues has automatically higher priority than language set for incoming calls.
How to add a new language to the PBX
If the language sets (voice records) provided in Kerio Operator do not satisfy your needs, you can download or buy different language sets and import them to the PBX. You can download the language sets (free or paid), for example, in the following sites: http://www.voip-info.org/ http://downloads.asterisk.org/
You can extract any language set archive and create your own voice records (provided you keep the file structure).
To add a new language:
1.
In the Kerio Operator administration interface, go to
Configuration > Advanced Options > Telephony
.
2.
Next to the
Default phone language
field, click
Configure
.
3.
In the
Voice Prompts
dialog box, click
Add
.
4.
Type a name of the voice prompt.
5.
Select a language syntax.
6.
Click
Upload
and select your sound package.
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7.
Click
OK
and click
Close
.
8.
In the
Default phone language
field, select the new language.
9.
Click
Apply
.
Changing indication tones according to countries
Each country has different standards for indication tones during calls (e.g. beeps, ringing tones, etc.).
You can change the settings in the administration interface.
To select a default country for your PBX, go to
Configuration > Advanced Options > Telephony
.
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Example
Live And Let Laugh Inc has the following configuration:
Joan Giggle, receptionist and operator, uses extension 100 and wishes the phone to communicate with her in the
Happish language.
lish.
Brian Snigger, receptionist and operator, uses extension 200 and is satisfied with the default language, which is Eng-
Phoney VoIP, an interface for incoming calls, is configured in Kerio Operator with the default language — English.
This interface is operated by Brian Snigger.
Telephium VoIP, an interface for incoming calls, is configured in Kerio Operator for communication with customers from the Bliss Seekers Land (in Cravish). This interface is operated by Joan Giggle.
The default language in Kerio Operator is English.
Voicemail is enabled and the extension for accessing the voicemail is 50.
Scenario 1:
When Brian Snigger calls Joan Giggle (200 > 100) or when Brian Snigger calls the voicemail (200 > 50), the automatic announcements are in English.
Scenario 2:
When Joan Giggle calls Brian Snigger (200 > 100) or when Joan Giggle calls the voicemail (200 > 50), the automatic announcements are in Happish.
Scenario 3:
Customers calling via the Phoney VOIP interfaces will hear announcements in the default language (English).
Scenario 4:
Customers calling via the Telephun VOIP interfaces will hear announcements in Cravish.
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4.7.2 Configuring Built-in DHCP server in Kerio Operator
Kerio Operator includes a built-in DHCP server. There are deployment scenarios in which it is useful to have a separate
DHCP server for VoIP devices:
In larger networks, you may need a
LAN segment dedicated to voice traffic .
In smaller networks, the router/firewall sometimes does not support the DHCP option 66 for automatic provisioning of phones.
WARNING
DHCP server is disabled in the default mode so that it does not collide with your existing DHCP server.
Configuring DHCP server
The built-in DHCP server must have a static IP address:
1.
In Kerio Operator administration interface, go to the
Network
section.
2.
Select a network interface and click
Edit
.
3.
In the
Interface Properties
dialog, switch configuration to
Use the following configuration
and type a new static IP address, mask and gateway.
4.
Check
Enable DHCP server
.
5.
Click
OK
to save the settings.
Kerio Operator will derive the configuration of the DHCP server from the values you set for the interface’s IP address, network mask, and gateway. The DHCP server sends option 66 to Kerio Operator’s own address with every address lease.
Assigning IP addresses
Kerio Operator generates the range of IP addresses from a configured mask of a network interface and assigns these addresses automatically.
Example:
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The configured IP address for a network interface is
192.168.62.1
The configured mask is
255.255.255.0
The gateway has the address
192.168.62.254
In this example, the range of IP addresses is
192.168.62.2 — 192.168.62.253
.
Example — LAN segment is dedicated to voice traffic
In our example, you have LAN and you need to add an other network interface as a special telephony segment.
Screenshot 56: DHCP is running in the particular segment —scheme
You need to configure two interfaces in Kerio Operator administration interface:
1.
Go to section
Configuration > Network > General
.
2.
Configure interfaces as displayed bellow.
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Screenshot 57: DHCP is running in the particular segment
4.7.3 Configuring parameter 66 in DHCP server in Kerio Control
The DHCP protocol assigns IP addresses. Apart from these addresses you can also send additional parameters via the
DHCP protocol. Parameter 66 configures the TFTP server address.
How to set parameter 66 in Kerio Control
1.
In the administration interface, go to section
DHCP server
.
2.
If you use the automatically generated scopes, use
Click to configure scopes manually
.
3.
Select a scope and open its settings (the
Edit Scope
dialog).
4.
Click on the
Add
button.
5.
Add parameter 66.
6.
Enter a public IP address through which Kerio Operator communicates.
4.7.4 Configuring server date, time and time zone in Kerio Operator
Time Settings
Correct time and time zone settings of your PBX are necessary for correct configuration of telephone communication, time ranges and logs. If the time zone is not set properly, log messages or call history may contain confusing information.
Therefore Kerio Operator is automatically synchronized with an NTP server.
WARNING
Do not change the settings unless you have a good reason.
NTP (Network Time Protocol) is a protocol for synchronizing time in your computer with time of the NTP server.
NOTE
Time and time zone settings on this tab refer to the administration interface time. It is the server time. Kerio Phone will display the time zone using the computer settings. If users are in a different zone to Kerio Operator, logs in call history will be displayed in users' time zone.
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Configuring synchronization with NTP
1.
In the administration interface, go to section
Advanced Options > General
.
2.
Select the
Keep synchronized with NTP servers
. Date and time can be set manually but it is better to use an NTP server which provides information about the current time and allows automatic management of the firewall's system time.
3.
Kerio Technologies offers the following free NTP servers for this purpose: 0.kerio.pool.ntp.org, 1.kerio.pool.ntp.org,
2.kerio.pool.ntp.org and 3.kerio.pool.ntp.org.
4.
Click
Apply
.
Screenshot 58: Advanced Options — date and time settings
Configuring time zone
1.
In the administration interface, go to section
Advanced Options > General
.
2.
Select a time zone from the
Server time zone
list.
3.
Click
Apply
.
The current date and time will be changed according to the new time zone.
4.7.5 Configuring standard phone interfaces
You can connect Kerio Operator to your provider using hardware cards.
You can use the card distributed with Kerio Operator Box series 3000 or you can use your own card and connect it to your
Kerio Operator server.
Supported cards
Kerio Operator supports the following cards:
PRI card — The number of concurrent calls varies depending on whether you have a contract with an American or
European provider:
T1 (in the USA) allows 23 concurrent calls.
E1 (in the EU) allows 30 concurrent calls.
BRI card — Has four ports, each of which can operate two concurrent calls.
FXO card — Has four ports each of which can operate only one call at a time.
For a specific list of supported cards, see the
Supported Phone Cards section on the Kerio website.
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Prerequisites
Before you configure an interface, you need to know:
Telephone number (or numbers) from your telephone provider
(PRI/BRI only) Which ISDN type to use for communication. This usually differs by your location: for example, EuroISDN for the EU, Nation ISDN Type 2 for the USA, and so on)
Whether your provider requires
Whether the provider sends or requires whole or abbreviated telephone numbers.
For more information, refer to
Mapping external and internal numbers (page 196).
At least one internal extension defined in Kerio Operator (for example, the extension of an employee who redirects the calls).
Configuring interfaces
After connecting a card, configure the interface:
1.
In the administration interface, go to the section
Configuration > Call Routing
. The
Interface and routing of incoming calls
table shows one of the following, depending on your card:
PRI card: one standard telephone interface
BRI or FXO card: four interfaces (one for each of the four ports)
2.
Double-click an unconfigured interface. The configuration wizard opens.
3.
Type a name for the interface (for example, your provider's name). The name must not contain spaces or special characters and must be unique.
One or multiple numbers
1.
If you acquire one or multiple phone numbers from your provider, type the numbers in the
New provider > With external number
field. You can:
Separate individual numbers with commas (for example, 555450, 555451, 555452, and so on)
Type the whole range using
-
(for example, 555450-555459)
2.
Click
Next
.
3.
Select an extension to receive all calls from the provider.
4.
(Optional) In the
Prefix to dial out
field, type a prefix for outgoing calls. Kerio Operator uses the prefix to route calls to your provider. This prefix can be same for other providers.
For more information, refer to Working with prefixes for outgoing calls (page 202).
5.
Click
Next
.
6.
(PRI and BRI only) Select the
Switch type
in the dialog box:
If you are in the EU, select the EuroISDN option
If you are in the USA, select the National ISDN Type 2 option
7.
Click
Next
.
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8.
Verify the information in the
Summary
section. If you need to add more information from your provider select the
Edit details of created interface
option.
For more information, refer to Configuring additional details for an interface (page 271).
9.
Click
Finish
.
10.
Create a rewriting rule to correctly
map numbers to internal user extensions
Interval of numbers
1.
If you acquire a trunk with an interval of numbers from your provider, type the numbers in the
New provider > With external number
field. Use x in place of the numbers that vary (for example, 555xxx).
2.
Click
Next
.
3.
Select an extension to which you want Kerio Operator redirect all calls to unassigned (unused) extensions.
4.
(Optional) In the
Prefix to dial out
field, type a prefix for outgoing calls. Kerio Operator uses the prefix to route calls to your provider. This prefix can be same for other providers.
For more information, refer to Working with prefixes for outgoing calls (page 202).
5.
Click
Next
.
6.
(PRI and BRI only) Select the
Switch type
in the dialog box:
If you are in the EU, select the EuroISDN option
If you are in the USA, select the National ISDN Type 2 option
7.
Click
Next
.
8.
Verify the information in the
Summary
section. If you need to add more information from your provider select the
Edit details of created interface
option.
For more information, refer to Configuring additional details for an interface (page 271).
9.
Click
Finish
.
10.
Create a rewriting rule to correctly
map numbers to internal user extensions
Overlap dialing
Some telephone providers require telephone numbers as a whole, others require the telephone numbers one digit at a time. Ask your provider about their requirements. Follow these steps to configure the interface:
1.
In the administration interface, go to the section
Configuration > Call Routing
.
2.
Select an interface and click
Edit
. The
Edit External Interface
dialog opens.
3.
Go to
Interface Card
.
4.
Select the
Overlap dialing
option.
5.
Click
OK
.
Configuring additional details for an interface
To set additional settings in your interface for incoming and outgoing calls:
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Select an interface and click
Edit
.
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3.
On the
Interface Card
tab, change the interface settings. See the following chapters for details.
4.
Click
OK
.
NOTE
If you select the
Edit details of the created interface
option on the last page of the
interface configuration wizard
, this dialog box displays automatically.
Adjusting audio gain for standard phone interfaces
NOTE
New in Kerio Operator 2.4!
To adjust audio gain:
1.
In
Configuration > Call Routing > Interfaces and routing of incoming calls
, select an interface and click
Edit
2.
Go to the
Interface Card
tab.
3.
Set
Rx gain [db]
.
4.
Set
Tx gain [db]
.
5.
Click
OK
.
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Configuring Type of number (TON)
NOTE
New in Kerio Operator 2.4!
Some providers send a stripped number with additional information about the type of the number. Kerio Operator can read these types and assign a prefix to the stripped number.
To configure prefixes for
Type of number (TON)
:
1.
In
Configuration > Call Routing > Interfaces and routing of incoming calls
, select an interface and click
Edit
.
2.
Go to the
Interface Card
tab.
3.
Click
Configure
next to
Type of number (TON)
.
4.
Type the prefixes you want to set.
5.
Click
OK
.
Increasing sensitivity of the DTMF detection
NOTE
New in Kerio Operator 2.4!
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To enable this option:
1.
In
Configuration > Call Routing > Interfaces and routing of incoming calls
, select an interface and click
Edit
.
2.
Go to the
Interface Card
tab.
3.
Select
Increase sensitivity of the DTMF detection
.
4.
Click
OK
.
Mapping of numbers
For more information, refer to Mapping external and internal numbers (page 196).
4.7.6 Connecting Kerio Operator to directory service
Which directory services are supported in Kerio Operator
Kerio Operator supports the following directory services:
Microsoft Active Directory
Apple Open Directory
What is the connection used for
In practice, mapping accounts from a directory service provides the following benefits:
Easy account administration
Apart from the internal database of user accounts, Kerio Operator can also import accounts and groups from an LDAP database. Using LDAP, user accounts can be managed from a single location. This reduces possible errors and simplifies administration.
Online cooperation of Kerio Operator and directory service
Additions, modifications or removals of user accounts/groups in the LDAP database are applied to Kerio Operator immediately.
Using domain name and password for login
Users may use the same credentials for Kerio Phone login and domain login.
WARNING
Mapping is one-way only, data are synchronized from directory service to Kerio Operator. Adding a new user in Kerio
Operator createsa local account — it will not be duplicated into the directory service database.
When creating user accounts in a directory service, ASCII must be used to specify usernames. If the username includes special characters or symbols, user may not be able to login to Kerio Phone or the administration interface.
If you disable users in Microsoft Active Directory, they are also disabled in Kerio Operator (they will not be able to login to Kerio Phone, make or receive calls with their extensions).
If you disable users in Apple Open Directory, they stay enabled in Kerio Operator.
Phone extensions can be managed in a directory service (if available) or locally in Kerio Operator. Select the most convenient option.
Connecting to a directory service
To map users from a directory service:
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Connect to directory service in section
Integration > Directory Service
.
.
All information about directory services can be found in the
Config
log.
Microsoft Active Directory
In the administration interface, go to
Integration > Directory Service
.
1.
Check the
Map user accounts from a directory service
option and select your directory service type.
2.
In the
Domain name
field, enter the name of your Microsoft Active Directory domain — the domain name is then copied in other necessary fields.
3.
In the
Hostname
field, enter the DNS name or IP address of the Microsoft Active Directory server. If you have a backup server, enter its name in the
Secondary hostname
filed.
4.
In the
Username
and
Password
fields, enter the authentication data of a user with at least read rights for Microsoft
Active Directory database. Username format is user@domain
.
5.
Within the communication of the Microsoft Active Directory database with the PBX, sensitive data may be transmitted
(such as user passwords). For this reason, it is recommended to secure such traffic by using SSL. To enable LDAPS in
Microsoft Active Directory, it is necessary to run a certification authority on the domain controller that is considered as trustworthy by Kerio Operator.
6.
The rest of the items in the dialog are completed automatically. Do not change them unless you have a special reason to do so. These items are Microsoft Apple Open Directory domain name and Kerberos Realm which has to match the
Microsoft Active Directory domain name, written in capital letters.
Apple Open Directory
In the administration interface, go to
Integration > Directory Service
.
1.
Check the
Map user accounts from a directory service
option and select your directory service type.
2.
In the
Domain name
field, enter the name of your Apple Open Directory domain — the domain name is then copied in other necessary fields.
3.
In the
Hostname
field, enter the DNS name or IP address of the Apple Open Directory server. If you have a backup server, enter its name in the
Secondary hostname
filed.
4.
In the
Username
and
Password
fields, enter the authentication data of a user with at least read rights for Apple Open
Directory database. Username format is: uid=root,cn=users,dc=domain,dc=tld
.
5.
Within the communication of the Apple Open Directory database with the PBX, sensitive data may be transmitted
(such as user passwords). For this reason, it is recommended to secure such traffic by using SSL. To enable LDAPS in
Apple Open Directory, it is necessary to run a certification authority on the domain controller that is considered as trustworthy by Kerio Operator.
6.
The rest of the items in the dialog are completed automatically. Do not change them unless you have a special reason to do so. These items are Apple Open Directory domain name and Kerberos Realm which has to match the Apple Open
Directory domain name, written in capital letters.
Activating users from a directory service
Once the mapping is set, select individual users and map them to the PBX. This is how to map users:
1.
Open the
Configuration > Users
section.
2.
Click
Import > Import from a Directory Service
.
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3.
In the dialog, select all users you wish to map (you can also add users later) and click
Next
.
4.
If users in the directory service have phone extensions assigned, you can either keep them or disable them. If you disable them, you have to assign new extensions. You can do it, for example, while changing your dial plan.
5.
Click on
Finish
. Activated users are displayed in section
Configuration > Users
.
NOTE
Only extensions in attributes telephoneNumber
(Microsoft Active Directory, Apple Open Directory) and otherTelephone
(Microsoft Active Directory) can be mapped (are displayed). If you create special attributes in a directory service for your phone numbers, you will not be able to map such extensions.
4.7.7 Connecting multiple Kerio Operators
In Kerio Operator, you can connect multiple Kerio Operator servers. This enables you to directly reach remote phones by their extensions for free and send or receive external calls through a relay server.
The section below describes how to connect these two servers:
The main server, which has internal extensions 100 — 199
The branch server, which has internal extensions 300 — 399
NOTE
For more information about routing of calls between Kerio Operator servers and the PSTN, see
Routing calls between multiple Kerio Operators and the PSTN .
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Prerequisites
Before the start of the configuration, you need:
Two Kerio Operator servers up and running
Extension schemes for both phone networks, each of which has a unique set of extensions
Both servers with public IP addresses or connected to the same network with a VPN tunnel
Connecting servers
On each Kerio Operator server, add a SIP interface for the other server.
1.
In the administration interface, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Click
Add SIP interface
. The
Add SIP Interface
dialog box opens.
3.
Type a name for the interface and select
Link to another PBX (without an external number)
.
4.
Click
Next
.
5.
In the
Prefix to reach the other PBX
field, type the appropriate number:
On the main server, type the prefix
3
(the first digit of each extension on the branch server)
On the branch server, type the prefix
1
(the first digit of each extension on the main server)
6.
Click
Next
.
7.
In the
Domain (IP address/hostname)
field, type the domain or the IP address:
On the main server, type the IP address of the branch server.
On the branch server, type the IP address of the main server.
8.
Disable the
Required to register
option.
9.
Click
Next
.
10.
Verify the information in the
Summary
section.
11.
Click
Finish
.
After the configuration of interfaces, Kerio Operator creates incoming and outgoing routes that use configured prefixes.
These routes do not rewrite any numbers. Make test calls between the connected servers to reach their extensions.
Example of a test call
Call number 305 from extension 110 on the main server:
1.
The user with an extension
110 dials number
305
.
2.
Kerio Operator on the main server recognizes the prefix
3 and routes the call to the branch server.
3.
The call arrives at the branch server and rings on the
305 extension.
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4.7.8 Routing calls between multiple Kerio Operators and the PSTN
NOTE
Redesigned in Kerio Operator 2.4!
Learn how to:
Reach the public switched telephone network (PSTN) from your connected Kerio Operator servers
Route incoming calls from the PSTN to your branch servers
NOTE
For more information about connecting multiple Kerio Operator servers, see
Connecting multiple Kerio Operators
.
The sections below use the following example:
Two connected Kerio Operator servers up and running:
The main server, which has internal extensions 100 — 199
The branch server, which has internal extensions 300 — 399
Outgoing calls from the branch server to the PSTN go through the main server.
Incoming calls from the PSTN to the branch server go through the main server.
The prefix for outgoing calls to the PSTN is 0.
External numbers from the SIP provider have the format 555 5xxx.
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Calling to the PSTN through the main server
To call to the PSTN via the interface of the main server:
Configure the interface on the main server.
Create an outgoing route on the branch server.
In the administration interface of the main server:
1.
Go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Double-click the interface for the branch server.
3.
Go to the
Advanced
tab.
4.
Select the
Allow incoming calls to use outgoing routes
option.
5.
Click
OK
.
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In the administration interface of the branch server:
1.
Go to
Configuration > Call Routing > Routing of outgoing calls
.
2.
Click
Add
.
3.
Type the prefix for outgoing calls of the main server (
0 in our example)
4.
Select the interface of the main server.
5.
Click
OK
.
Make a test call to reach a number in the PSTN from the branch server.
Example of a test call
Call 555-9999 from extension 305:
1.
The user with an extension
305 dials the number
0-555-9999
.
2.
Kerio Operator on the branch server recognizes the prefix
0 and routes the call to the main server.
3.
The call arrives at the main server.
4.
Kerio Operator on the main server recognizes the prefix
0 and strips the prefix off.
5.
The main server routes the call to the SIP provider.
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Routing incoming calls from the PSTN to the branch server
To route incoming calls to the branch server:
If you have separate numbers, use speed dial extensions
If you have a trunk of numbers, rewrite called numbers to match the internal extensions of the branch server
Using speed dial extensions
To use speed dial extensions:
1.
Create a speed dial extension (
9305 in the example) that dials an extension of the branch server (
305 in the example). For more details, see
2.
Go to
Call Routing
and double-click the interface for the branch server.
3.
Go to the
Advanced
tab.
4.
(Optional) To enable users to return calls, select
Do not substitute the calling number when forwarding calls
and click
OK
. This option also displays the caller ID of the caller instead of the number of the speed dial extension.
5.
Double-click the number from your provider that you want to map.
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6.
In the
Route incoming calls to
field, select the speed dial extension and click
OK
.
7.
Repeat steps 1—6 for all extensions you want to map to the branch server.
From now on, Kerio Operator uses the speed dial extension for all incoming calls that reach the external number and routes the call to extension
305 of the branch server.
Make a test call to reach an extension on the branch server.
Example of a test call
Call 555-5305 from 555-9999:
1.
Caller dials the number
555-5305
.
2.
The call arrives at the main server.
3.
Kerio Operator routes the call to the
9305 extension and then to
305
.
4.
The main server recognizes the prefix
3 and routes the call to the branch server.
5.
The call arrives at the branch server and rings on the
305 extension.
Using number rewriting
To rewrite called numbers from your trunk and route them to the branch server:
1.
In the administration interface of the main server, go to
Configuration > Call Routing > Interfaces and routing of incoming calls
.
2.
Double-click the interface for the branch server.
3.
Go to the
Advanced
tab.
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4.
(Optional) To enable users to return calls, select
Do not substitute the calling number when forwarding calls
and click
OK
. This option also displays the caller ID of the caller instead of the number of the speed dial extension.
5.
Double-click the interface for your provider.
6.
Go to the
Advanced
tab.
7.
Enable the
Allow incoming calls to use outgoing routes
option.
8.
Double-click the trunk of numbers to verify that Kerio Operator rewrites the called number correctly.
9.
Click
OK
.
After configuring the interface, make a test call to reach an extension on the branch server from the PSTN.
Example of a test call
Call 555-5310 from 555-9999:
1.
Caller dials the number
555-5310
.
2.
The call arrives at the main server.
3.
Kerio Operator matches the call to a SIP interface and strips off the first four digits of the number.
4.
The call automatically uses the outgoing route with the prefix
3 and arrives to the branch server.
5.
The call rings on the
310 extension.
4.7.9 Creating and using speed dial
Speed dial is a shortcut for phone numbers (for both the internal extensions and external phone numbers).
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Adding speed dial
Before you begin creating speed dial, select a numerical range you will use. Speed dial must be different from current extensions. Generally, it is convenient to create speed dial so that they will not coincide with your dial plan in future.
1.
Open
Speed Dial
.
2.
Click
Add
.
3.
In the
Add Speed Dial
dialog box, type a speed dial in the
Speed dial extension
field.
4.
In
Dial number
, type the callee's phone number including the prefix for outbound calls.
5.
Click
OK
.
Configuring speed dial with DTMF
The speed dial with DTMF (Dual-tone multi-frequency signaling) is intended for calling special services like long distance phone service providers. If you need to place a call via such a service, you usually need: provider's number (usually it is a toll-free number that starts with
800
: 800555333) user ID (78901234)
PIN (8808) a number you want to call (011420111222333)
# character — denotes the end of the number and starts the call.
When you set the speed dial with DTMF, the number of steps is shortened to dialing the speed dial extension followed by the number you want to call: 89011420111222333
Configuring speed dial with DTMF
1.
In the administration interface, go to
Speed Dial
.
2.
Click
Add
.
3.
In the
Add Speed Dial
dialog box, type a speed dial in the
Speed dial extension
field. In our example it is
89
.
4.
In
Dial number
, type the provider's access number including the prefix for outbound calls.
5.
Select
DTMF tones are enabled
. Once you enable DTMF, the speed dial behaves as a dial-out prefix.
6.
In the
DTMF prefix
field, type the access code and PIN. Your provider's IVR system may require a pause between typing the access code and PIN. Therefore use the w character for a half second pause. In our example it is
78901234w8808
.
7.
In the
DTMF suffix
field, type
#
8.
Click
OK
.
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Screenshot 59: Add Speed Dial dialog
What is happen if you use the speed dial 89?
You want to call the number 011420111222333.
To place the call, you dial: 89011420111222333. The service will dial the access number 800555333 and once the call is connected, the following DTMF digits are sent:
789012348808 011420111222333 #
4.7.10 Creating ringing groups
You can use ringing groups to make calls ring simultaneously on multiple extensions.
Adding new ringing groups
1.
In the Kerio Operator administration interface, go to
Ringing Groups
and click
Add
.
2.
In the
Group extension
field, type the extension number for the group.
3.
Add extensions you want to ring simultaneously to the table.
4.
(Optional) To redirect the call to another extension when no one answers the phone, select
Fall back to another extension when the group is not responding
and set a timeout and destination extension.
5.
(Optional) If you don't want to display the answered call as missed on other phones in the group, go to
Advanced
tab and select
Do not display missed calls on the phones
.
6.
Click
OK
.
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4.7.11 Customization of voice sets
This summary provides information on how to customize/change voice sets in Kerio Operator.
The Internet provides many sources of localized and customized basic sounds and voice prompts. Voice sets for various languages can be found at http://www.voip-info.org
. However, it is recommended to use voice sets present in Kerio
Operator than downloading them from the Internet, as it may not include all of the prompts used by Kerio Operator.
To customize voice sets:
1.
Log in to Kerio Operator administration.
For more information, refer to Logging into Kerio Operator Administration (page 19).
2.
Go to the
System Health
dialog, hold the
shift
key and click
Tasks
. You should see an option to enable
SSH
.
3.
Log in to your system with an SSH client (using root and your admin password).
4.
Go to
/var/lib/asterisk/sounds
.
5.
Download the directory of sounds you will use as a base (e.g. en_GB).
6.
Modify the .gsm sound files using any compatible sound editor.
7.
Compress the folder.
8.
Back in the Admin Console, open
Advanced Options > Telephony
.
9.
Click
Configure
next to
Default phone language
option and upload the voice sets.
Once you upload a voice and sound set, you can use it for Kerio Operator, individual interfaces or individual users.
For more information, refer to Language settings in Kerio Operator (page 261).
4.7.12 Customizing the Kerio Phone login page
Adding your custom logo
To change a logo of your login page:
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1.
In the administration interface, go to
Configuration > Advanced Options > Login Page
.
2.
Select the
Use custom logo on login page
option.
3.
Click
Change
and locate the new logo file. The logo must be in the PNG format. The recommended maximum size is
325 x 80 pixels.
4.
Click
Apply
to save your settings.
Configuring your custom button style
To change a style of a button:
1.
In the administration interface, go to
Configuration > Advanced Options > Login Page
.
2.
Select the
Use custom button style
option.
3.
Type a color's hex value for
Text color
(for example,
#ffffff
).
4.
Type a color's hex value for
Background color
(for example,
#669900
).
5.
Click
Apply
to save your settings.
Adding your custom text
To add a text to your login page:
1.
In the administration interface, go to
Configuration > Advanced Options > Login Page
.
2.
Select the
Add the following text to the page (supports HTML)
option.
3.
Type your text (for example,
In case of emergency issues, call 555-1234
).
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4.
Click
Apply
to save your settings.
4.7.13 Distinctive ringing support
Kerio Operator supports setting different ring tones for different types of calls (external calls, internal calls or ringing groups).
Configuring strings Kerio Operator
NOTE
New in Kerio Operator 2.5!
By default, Kerio Operator uses the following strings for the Alert-Info header:
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operator-external
(calls from an interface) operator-queue
(calls from a call queue) operator-group
(calls to a ringing group)
To configure different ring tones for your SIP interfaces, call queues and ringing groups, change the default string in
SIP
"Alert-Info" header
:
1.
Go to
Call Routing
, or
Call Queues
, or
Ringing Groups
.
2.
In
Call Routing
, double-click a SIP interface. In
Call Queues
or
Ringing Groups
, double-click an extension.
3.
Switch to the
Advanced
tab.
4.
In
SIP "Alert-Info" header
, change the default string.
5.
Click
OK
.
Configuring telephones (example: snom 360)
1.
Go to web administration of your telephone.
2.
Go to
Setup > Preferences
3.
Find the alert-info settings.
4.
Set different ringers for different alert-info strings (see screenshot).
5.
Save the settings.
For testing purposes: Try to make a call from an external telephone number, from an internal extension and to ringing group.
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Screenshot 60: Customising Alert-Info strings
4.7.14 Fax support in Kerio Operator
Using fax in Kerio Operator
Kerio Operator supports:
T.38 support
T.38 is a protocol for realtime transmission of fax over IP.
Kerio Operator uses T.38 by default. Ask your provider whether they support this protocol. If not, read section
My provider does not support T.38
Connecting a fax machine to Kerio Operator
1.
Connect your fax machine to an Analog Telephone Adapter device (ATA — for example, Cisco SPA 112).
2.
Assign one of Kerio Operator extensions to the ATA device.
Fax machine is connected to the network. You can send and receive faxes.
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Configuring an ATA device
You can use various ATA devices. Each device has different settings. The following must be configured:
1.
enable T.38
2.
set
fax passthru
to
ReInvite
NOTE
in Kerio Operator sets these variables automatically.
Receiving faxes to a user's email address
You can enable fax-to-email service for any extension. Kerio Operator then sends all incoming faxes to the user’s email address as PDF attachments.
WARNING
In the administration interface, define SMTP relay in section
Advanced Options > General
so that your Kerio
Operator can send emails.
In the administration interface: go to
Users
and enter an email address for each user.
go to
Extensions
and enable option
Forward incoming faxes to user's address
for the particular user's extension.
Configuring fax detection (CNG signal)
A CNG signal is the fax machine sound you may hear when there is a fax machine connected to the other end of line.
Kerio Operator can detect the signal and start receiving faxes automatically.
1.
In the administration interface, go to
Extensions
.
2.
Double-click a selected extension.
3.
On tab
Advanced
, select:
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Dial extension, wait for answer, detect CNG
— PBX dials an extension, waits for an answer and then starts detecting the CNG signal. User has to answer a call first in order to receive faxes. When a fax tone is detected, the call will be taken over by Kerio Operator.
Answer, detect CNG, dial extension
— PBX answers a call first, then detects the CNG signal and immediately dials an extension. If users don't answer the phone, a fax mail is received and users have a missed call on their phone display. This option is good for occasional fax transmissions.
Answer, detect CNG, wait 3.5 seconds, dial extension
— Extension is dialed after a 3.5 seconds delay which is used to detect faxes. There will not be any missed calls shown on the phone's display. Regular calls will be automatically answered and will be followed by a 3.5 second delay of silence. This option is good for more frequent fax usage.
Answer, detect CNG, wait 3.5 seconds (ringing tone), dial extension
— the PBX will generate a ringing tone instead of waiting in silence. This option is also good for more frequent fax usage and may be less confusing to human callers.
4.
Save the settings.
Receiving all faxes to a specific email address
Kerio Operator can send all incoming faxes to a single email address.
1.
Go to
PBX Services
.
2.
Open
Receive fax messages
.
3.
Type email address in the
Send received faxes by email to
field.
Kerio Operator will send all incoming fax messages to the specified email address.
My provider does not support T.38
WARNING
Fax support without T.38 is not reliable. Using codecs G.711 A-law/U-law instead of T.38 is a workaround.
If your SIP provider does not support T.38, you have to solve these issues:
Enable codecs G.711 A-law/U-law for the transmission. High compression codecs would distort signal.
Reduce the speed on your fax machines (if supported).
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Enabling G.711 A-law/U-law codecs for the interface
1.
Login to the administration interface.
2.
Go to
Configuration > Call Routing
.
3.
Click the provider's interface.
4.
Click the
Codecs
tab.
5.
Move
G.711 A-law
and
G.711 U-law
to the
Selected codecs
table.
6.
Move
G.711 A-law
and
G.711 U-law
codecs up in the table.
NOTE
Moving G.711 A-law/U-law codecs up in the table can cause bandwidth consumption.
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7.
Click
OK
.
Fax messaging now uses codecs G.711 A-law/U-law.
Disabling the T.38 support
Although your SIP provider supports T.38 protocol, you may experience some difficulties in communication. Conclusion is disabling a support of the T.38 protocol:
1.
In the administration interface, go to
Configuration > Advanced Options
.
2.
On the
General
tab, click
Configure...
next to the
SIP Configuration
.
3.
Unselect
Use T.38 standard for faxing
.
For more information, refer to My provider does not support T.38
Sending PDF to fax
For more information refer to
Sending PDF to fax in Kerio Phone
.
4.7.15 Hosting Kerio Operator
In some situations, it may be preferred to deploy Kerio Operator at a remote site, or data center, as these locations may offer better bandwidth, reliability, and consolidated management. This topic addresses the considerations when deploying Kerio Operator outside of the local network.
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Phone Provisioning
When Kerio Operator is deployed on the same network as the IP phones, the provisioning process can be handled automatically through DHCP. However, if Kerio Operator is remote, the phones typically must be configured manually.
Consider pre-configuring the phones at a convenient location before deployment at the remote site. For ongoing maintenance of the phones, it will be necessary to use the web administration of the phones, which will require access to the remote network. We've found Snom, Yealink, and Linksys phones to offer the best options for web configuration and remote management. For added security, automatic provisioning should be disabled from the dialog in the web administration, located under Provisioned Phones > Phone Provisioning.
With the introduction of the Kerio Phone mobile app, configuration from mobile devices is quite easy, and ideally suited in case Kerio Operator is in a hosted, or remote environment.
For more information, refer to Provisioning of Kerio
Operator Softphone for mobile devices (page 175).
Network Configuration
Because Kerio Operator includes a built-in firewall, it's not necessary to incorporate an external firewall. If possible, Kerio
Operator should be assigned an Internet routable IP address to avoid network address translation, as this can cause a variety of call quality, or connectivity type issues. In many cases, the phones will be connecting through NAT, and there is an option in the extension properties to account for this.
For more information, refer to Configuring NAT (page 258).
Bandwidth
When Kerio Operator is remote to the phone network, it means that local calls between extensions will be routed through the Internet. It's important to take this into consideration when evaluating the bandwidth requirements for a remote network of phones. A typical phone call may consume about 80 kbps, although some codecs may consume less bandwidth. For locations with limited bandwidth, consider setting GSM as the preferred Codec for those extensions provisioned on that network.
For more information, refer to Bandwidth used by the different codecs (page 208).
Security
In a hosted scenario, Kerio Operator is typically accessible directly over the Internet (unless access is restricted through
VPN). It's therefore necessary to pay close attention to the security settings related to Kerio Operator.
For more information, refer to Securing Kerio Operator (page 254).
4.7.16 Integrating Kerio Connect and Kerio Operator
What are the possibilities of Kerio Operator and Kerio Connect integration
There are several posibilities how to integrate Kerio Operator and Kerio Connect:
Integrating voicemail
The integration synchronizes flags which marks whether a voicemail message has been read/played. If you mark a message as read in Kerio Phone or if the message is marked as read after you hear it on your phone, the message will also be flagged as read in your mailbox (and vice versa).
If integration with Kerio Connect is set, voicemail messages are not stored in Kerio Operator but in user's
Inbox
on the mailserver.
WARNING
Limitation: You can integrate Kerio Connect with a single Kerio Operator only.
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Searching the address book on Kerio Connect on provisioned phones
Calling directly from Kerio Connect Client
For more information, refer to Configuring Click to Call in Kerio Connect client (page 302).
Configuring voicemail integration
If you want to set up voicemail integration, follow these steps:
1.
Go to
Configuration > Users
.
2.
In the users' settings, type their email addresses.
WARNING
Use the primary email address (not an alias) — otherwise sending of messages to Inbox will not work.
3.
Go to
Configuration > Voicemail > Email
.
4.
Change the SMTP server settings to
Integrate with Kerio Connect
.
5.
Click
Configure
and type the DNS name of Kerio Connect.
NOTE
If the IMAP service runs on a nonstandard port in Kerio Connect, enter the server name including the port number
( hostname:12345
)
6.
Specify the name and password of a user with admin rights for Kerio Connect.
Authentication details are used for the first connection to Kerio Connect and creation of a special account using JSON-
RPC2 API for authentication. Once this special account is created, the PBX drops the administrator's name and password.
NOTE
To synchronize flags between the two servers, Kerio Operator uses protocol IMAP with TLS or IMAPS. If Kerio Connect is behind firewall, enable at least one service on standard port. The IMAP or IMAPS services need to be allowed on
Kerio Connect server.
Opening ports
If servers are behind a firewall, open the following ports:
143/993
4040
Troubleshooting
If Kerio Connect is protected by firewall, verify that the 143/993 and 4040 ports for the IMAP/IMAPS protocols are open.
The IMAP/IMAPS services must be running in Kerio Connect.
If you cannot connect Kerio Operator with Kerio Connect, consult the following logs:
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In Kerio Operator, consult the Warning log for any problems with the IMAP service.
In Kerio Operator, consult the Error log for problems with connection to Kerio Connect’s IMAP server.
In Kerio Connect, consult the Mail log for information about delivered voicemails.
4.7.17 Setting optional call recording
Call Recording
Call recording is subject to special laws in many countries and may not be legal in your jurisdiction, or may require notice to the other party of the call.
In Germany and maybe in other countries you have to give callers (your customers) an option to continue without having their call recorded. This topic helps you with the following simple settings:
Creating two call queues, one with call recording, the second without recording.
Preparing voice prompt (something like "This call is recorded ..."). How to easily create voice files you can find out in
topic.
Creating simple auto attendant script which allows your customers to choose call queue without call recording.
Recorded calls backup.
How to create and configure call queues
1.
In the administration interface, go to section
Configuration > Call Queues
.
2.
Open the
Add Call Queue
dialog and enter the extension number for the new queue on tab
General
(for example
400).
3.
Description could be Calls are recorded.
4.
Click on
Record calls
.
5.
Set other parameters according to the
topic.
6.
Open the
Add Call Queue
dialog again and enter another extension number on tab
General
(for example 401).
7.
Description could be Calls are not recorded.
8.
Set remaining parameters just as the first call queue.
9.
Do not click on
Record calls
.
NOTE
Agents need to logged to both queues.
Now, you have two equal call queues. One of them is recorded and the other is not recorded.
How to create auto attendant script for two call queues
1.
In the administration interface, go to
Configuration > Auto Attendant Scripts
.
2.
Click
Add
.
3.
Enter script extension (for example 500).
4.
In the
Edit Menu
dialog, click
Edit
.
5.
Add some description (script for recorded and unrecorded call queues).
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6.
Select prepared anouncement (for example: "This phone call will be recorded. If you do not want recording your call, please, press 1 and wait for the connection.")
7.
Set
Default action
to
Dial extension number
and enter extension of the call queue with call recording.
8.
Create a new row by clicking on
Add
.
9.
Key is set to 1.
10.
Set
Action
to
Dial extension number
and specify the extension of the call queue without call recording.
11.
Click
OK
.
For more information, refer to Configuring auto attendant scripts (page 230).
Saving recorded calls
Kerio Operator can automatically backup recorded calls: locally — default settings via FTP
Saving recorded calls via FTP
1.
In the administration interface, go to section
Integration > Remote Storage
.
2.
Type a hostname of your FTP server
3.
Type a username and password if it is necessary.
4.
Test the connection.
5.
Click
Apply
.
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6.
In section
Status > Recorded Calls
, click
Settings
.
7.
Select
Save to remote storage
.
8.
Select a type to FTP.
9.
Type an FTP path.
Test the FTP path by clicking on the
Test Connection
button.
4.7.18 Setting outgoing calls constraints in Kerio Operator
You may want to limit some or all outgoing calls for a variety of reasons. For example, should an outside party obtain the username and password of one of your employees, they could use your PBX for international calls—possibly involving fraud and costing you money. It is therefore critical to have calls to external networks well configured.
You can set outgoing call constraints to prevent these types of attacks.
Restricting the length of individual outgoing calls
To set the maximum call duration:
1.
In the administration interface, go to
Configuration > Security
.
2.
Set
Maximum duration of each outgoing call
. The recommended value is 2 hours.
Restricting the number and length of outgoing calls
You can limit all outgoing calls by creating special rules in the section
Configuration > Security
in table
Outgoing calls constraints
.
The default rule limits the number of outgoing calls to 50 per hour and total call duration to 2 hours per day.
Example
A manufacturer in the United States sells and primarily has contacts just in the U.S. and Canada, but has a factory in
Mexico. Management wants to limit calls to other countries.
1.
In the administration interface, open
Configuration > Security
and click
Add
.
2.
Type a rule name, such as
Constraints for Mexico
).
3.
In the
Apply to these outgoing calls
section, select
All except listed
and click
Add
.
4.
Add the calling prefixes as a single string:
For local calls: 9 (outside line)
For U.S. and Canada: 91 (outside line + 1 preceding the area code)
For Mexico: 901152 (outside line + 011 for international call + 52 for Mexico's country code)
5.
Define the conditions: Set
Maximum calls count
to
10 per hour and
Maximum total calls duration
to 1 hour a day.
6.
When the conditions are met, Kerio Operator can send a warning email or block all outgoing calls.
We recommend creating:
One soft rule with lower limits that sends warning messages via email.
Another rule with higher limits that blocks the PBX.
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WARNING
If the limits are reached and the PBX is blocked, no one will be able to make calls to the restricted prefixes. However, an administrator can unlock the PBX in section
Configuration > Security
. We recommend making a thorough analysis of your calls before setting restrictions so that the PBX is not blocked by standard operations.
In addition to these settings, you can also configure similar rules for specific users or groups of users.
For more information, refer to Disabling outgoing calls to certain countries or regions (page 216).
4.7.19 Tips for Apple iPad
This topic provides a few useful tips for a better administration user experience on Apple iPad.
Screen orientation
It is recommended that iPad is held in the landscape mode while working with the Kerio Administration interface. For viewing longer dialog boxes, hold the device in the portrait mode.
Tree of sections
To get more space to view the section content, hide the tree of sections on the left.
Pop-up menu
To open context menu (e.g. in logs), tap the screen with two fingers at a time.
Sort by columns
Select the column and tap to set sorting or open a menu.
Editing table values
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First, select a table row. To change the value, single-tap the particular spot.
Logs
If you use search, you can go to the previous or next occurrence by using the arrow buttons.
Log pages can be scrolled by dragging with fingers. The more fingers you use, the faster the page scrolls.
NOTE
If you have Multi-Touch allowed on iOS 5, you can use up to three fingers for log scrolling.
4.7.20 Using paging groups and services
Paging, also known as intercom or public address, enables Kerio Operator users to broadcast a message to a user or a group using a phone's speakers. Phones included in the paging group or service answer the call automatically, and activate the loud speaker.
WARNING
Paging works with
phones that support auto-answer functionality
The paging group is a group of users to whom you can make a call with using loud speaker.
The paging service is a prefix for paging. You dial the prefix + an extension to page a particular user.
Configuring paging groups
1.
In the Kerio Operator administration interface, click
Paging
.
2.
Click
Add Group
.
3.
Type the paging group extension.
4.
To add members to the group, click
Add
.
5.
(Optional) Check
Page only idle extensions
. Paging does not interrupt active calls.
6.
(Optional) Check
Beep when the call is established
. Your phone beeps when all phones from the paging group are connected.
7.
Select audio transfer strategy: Select
only to the receiving party
to broadcast the message without giving paging group members ability to answer. Select
in both directions
to enable two-way communication.
8.
(Optional) To enable call recording, select
Record Calls
.
9.
Click
OK
.
If you want to check your configuration, dial the group extension and do a test call.
Configuring a paging service
1.
Go to the administration interface, and click
Paging
.
2.
Click
Add Service
.
3.
Type
Paging service prefix
.
4.
(Optional) Check
Page only idle extensions
. Paging do not interrupt active calls.
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5.
(Optional) Check
Beep when the call is established
. Your phone beeps when all phones from the paging group are connected.
6.
Decide, if you want to transfer audio
only to the receiving party
(telephones play the message and users cannot answer) or
in both directions
(telephones play the message and users can answer).
7.
(Optional) To enable call recording, select
Record Calls
.
8.
Click
OK
.
If you want to check your configuration, dial the service prefix and an extension and do a test call.
Securing paging
Anyone who knows the extension or whole telephone number of the paging group can use this feature. You can secure your paging groups and service with
Call Permissions
. You can create a new call permission group, where paging an extension or a prefix is denied and add people without permission for using paging:
1.
In the administration interface, go to
Definitions > Call Permission Groups
.
2.
Click
Add
.
3.
In the
Edit Call Permission Group
dialog, type a group name (for example
Paging
).
4.
Click
Add
.
5.
In the
Add Prefix
dialog, type a paging extension or service.
6.
Click
OK
.
7.
Go to
Configuration > Extensions
.
8.
Select the user who will have paging disabled and click
Edit
.
9.
In the
Call permissions group
menu, select the paging rule (in our example it is
Paging
).
10.
Repeat step 9 to disable paging for additional users.
For testing purposes you can add yourself to restricted group called
Paging
. Try to call the paging group or service.
List of supported and tested phones
Paging was tested by Kerio Technologies with the following telephones:
Cisco SPA508G, SPA525G
Linksys SPA942, SPA922
Polycom IP335, IP650
Well SIP-T38G
Snom 360, 820 and MeetingPoint
4.7.21 Configuring Click to Call in Kerio Connect client
New in Kerio Operator 2.3!
Users of Kerio Connect client can click a contact's phone number to initiate a call from Kerio Operator. By clicking a number, you can select the registered phone/device to dial from. The selected phone/device will ring. Answer the call and Kerio Operator will place the outbound call to the dialed number.
To set up and use the Click to Call feature in Kerio Operator, go to the
Configuring a number transformation
section.
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To set up and use the Click to Call feature in Kerio Connect, go to the Making calls from Kerio Connect topic.
If you want to Click to Call for Kerio Operator plugin for Chrome and Firefox, go to the
Using Click to Call for Kerio
Operator plugin for Chrome and Firefox topic.
Configuring a number transformation
Numbers dialed by Click to Call must be in the same format as for usual calls. If you use an outgoing prefix in your environment, you must add a number transformation rule to Kerio Operator:
1.
In the administration interface, go to
Integration
.
2.
On the
Number Transformation
tab, add the rule for your outgoing prefix (for example 9).
3.
Click
Add
.
4.
In the
Add Number Transformation Rule
dialog, type dot in the
Match number
field. Numbers of any length are matched.
5.
In the
Add prefix
field, add the outgoing prefix (for example 9).
6.
Click
OK
.
Screenshot 61: Rule for outgoing prefix
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5 Troubleshooting
This section helps you fix problems you might encounter when using Kerio Operator.
5.1 Common issues
This section provides solution to various issues:
5.1.1 Troubleshooting connections to SIP providers
5.1.2 Troubleshooting call quality issues
5.1.3 Browser extensions or add-ons may interfere with Kerio products
5.1.4 Cannot play voicemails or audio files in Safari
5.1.1 Troubleshooting connections to SIP providers
This topic describes what information you must acquire from your provider and offers tips on configuring SIP interfaces in Kerio Operator.
The sections below cover these topics:
For more information, refer to Connecting to VoIP service providers (page 25).
For more information, refer to Mapping external and internal numbers (page 196).
SIP SRV records
SRV (service) records are entries in DNS that specify the location of service servers. Some SIP providers have SIP SRV records defined for their domain name. Asterisk uses SIP SRV resolution for outbound calls.
To make outbound calls, you must add all proxy servers from your provider's SRV records to your Kerio Operator SIP interface.
To obtain proxies, you can:
Ask your provider directly.
Use a Linux dig command.
Example for nexvortex.com:
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Command: dig _sip_udp.nexvortex.com SRV
Result:
_sip._udp.nexvortex.com 1800 IN SRV vortex.com.
_sip._udp.nexvortex.com 1800 IN SRV px7.nexvortex.com.
_sip._udp.nexvortex.com 1800 IN SRV vortex.com.
20 0 5060 px5.nex-
30 0 5060
10 0 5060 px1.nex-
To add the proxy servers to the Kerio Operator SIP interface:
1.
In the Kerio Operator administration interface, go to
Configuration > Call Routing
.
2.
Double-click the SIP interface.
3.
Go to the
SIP Details
tab.
4.
Type the names of the proxy servers in the
Inbound proxy
field.
px5.nexvortex.com
, px7.nexvortex.com
, and px1.nexvortex.com
in the example above.
5.
Click
OK
.
Domains and usernames
If you have issues related to domain or user names while configuring a SIP interface:
Verify that the Authentication usernames and the SIP name are correct. If you don't get any Authentication username from your provider, assume that they are the same.
Verify your provider's domain name. Some providers use different terminology, for example, "server name" or "identifier to be used instead of the host name part of the SIP URI".
Verify that the provider uses the same server name for all SIP server roles (registration, inbound proxy, outbound proxy). If not, configure the Kerio Operator SIP interface correctly for your provider.
If your provider has multiple SIP servers, type all of them in the Kerio Operator SIP interface.
Phone numbers
Number formats
Before you configure a SIP interface and incoming and outgoing routes in Kerio Operator:
Verify the format of phone numbers your provider uses.
The specific number of digits, for example, 9-digit numbers, 10-digit numbers, and so on. note that some US providers use 11-digit numbers instead of 10-digit numbers. In that case, the first digit is always
1
.
E.164 number format, where numbers start with the
+ sign followed by a country code (for example,
+1 in the US). This format often requires configured number rewriting for outgoing calls. For example, rewrite dialed numbers that start with
9 so that the called numbers start with
+1
(
94084964500 to
+14084964500
).
Custom number format, where, for example, providers use the international format without the
+ sign.
Rewrite all national and international numbers to the custom format of the provider.
Verify that the number format for inbound calls and outbound calls is the same. If not, configure number rewriting correctly for your provider. For example, your provider sends you 10-digit numbers, but requires 11-digit numbers for
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outbound calls. Create an outgoing route that rewrites numbers to 11-digit format.
Phone numbers as identifiers
Your provider can use phone numbers instead of SIP usernames. If you have only a single external number, many providers use your external number as a SIP username as well.
If you have multiple numbers, your provider:
Uses one of the external numbers as the SIP username.
Gives you a random string or the common part of your external numbers.
Gives you multiple accounts, each account with a single external number. Create a SIP interface for each account. You can then assign all interfaces under the same dial-out prefix.
SIP headers
Call setup
When a device initiates a SIP call, it sends the SIP INVITE request. The beginning of the request looks like this:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.1.99:5060;branch=z9hG4bK343bf628;rport
From: "Test 15" <sip:[email protected]>tag=as58f4201b
To: <sip:[email protected]>...
In the request above, extension 15 calls extension 13.
The called number is in the INVITE (
Request-Line
in the Kerio Operator administration interface) and To headers.
The calling number is in the From header.
After you create a SIP interface, Kerio Operator reads the calling number from the From header and the called number from the INVITE header by default. Verify which headers your provider requires and change the settings in the Kerio
Operator SIP interface.
For more information, refer to Connecting to VoIP service providers (page 25).
Example
You have multiple numbers in range 5551200-5551299.
Your username is the common part of your numbers, 55512.
The SIP provider sends 55512 in the INVITE header and the specific number, for example 5551234, in the To header.
For outgoing calls, your provider requires 55512 in the From header and the calling number in the P-Preferred-Identity header.
Transferring calls
When transferring calls, Kerio Operator can notify the receiving party about the original caller, otherwise the callee cannot recognize the origin of the call.
Enable the
Diversion
header on the
SIP Details
tab of the SIP interface.
If the Diversion header doesn't work, ask your provider which SIP header to use.
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5.1.2 Troubleshooting call quality issues
When using voice over IP, there are many considerations which can affect the quality of phone calls, and identifying the source of the issue can sometimes be difficult. This topic is designed to describe the different types of call quality issues, the general causes, and a likely solution to each type of issue.
Symptom:
One-way, or No Audio
Cause:
Improper NAT settings, or improper handling of SIP/RTP by the firewall.
Solution:
Avoid NAT, Bypass the firewall by connecting Kerio Operator directly to the WAN, and attaching a second network interface to the local network. Otherwise, verify the NAT settings in Kerio Operator and the connected SIP device.
Configuration details are available in KB topic
. Also check the kerio forums or other resources for known
SIP handling issues with the brand of firewall in front of Kerio Operator. The firewall should be configured to allow
TCP/UDP port 5060 and 5061, and UDP ports 10000 ~ 20000
Symptom:
Garbled, or incomprehensible voice transmission.
Cause:
Bad Codec
Solution:
Identify and remove the bad Codec from the selected codecs in the call route.
NOTE
Troubleshooting tip: During a call, review Status > Calls and check the used Codec. Compare it to calls that don't have the issue.
Symptom:
Choppy voice quality. Some words or parts of the conversation are lost or cut off.
Causes:
Insufficient or jittery Internet bandwidth.
Local networking issue (bad cable, overloaded switch, bad NIC, overloaded WiFi)
High CPU usage or insufficient hardware resources
Solutions:
Set GSM as the only selected Codec on the call route. Research options for allocating more bandwidth or incorporating QoS.
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Update firmware of the phone, replace the cable to the phone, change the Codec of the extension, replace the phone.
Upgrade hardware to faster storage device (e.g. SSD) or add RAM
NOTE
Troubleshooting tips: Enable extension 81 for the echo test. Dial from various locations and phones to isolate the conditions under which the choppiness occurs. From a browser on that network, go to pingtest.net
and check the line quality. Check the status > system health to make sure the CPU/Memory is not overloaded. Check settings for call recording and logging to ensure reduced File I/O.
Capturing the network communication (packet capture or packet dump):
If the previously mentioned steps were not helpful, it may be necessary to capture the network communication for analysis. To do this, navigate to Configuration > Network. Select the network interface which is linked to the network where the issue is observed. Click the 'Packet Sniffer...' button to the right of the interfaces dialog. Start the capture, then attempt to reproduce the problem. Once the issue has been observed, stop the capture and download the capture file. If you have an open case with technical support, they may request this file.
5.1.3 Browser extensions or add-ons may interfere with Kerio products
When you have trouble working with an administration or client interface of Kerio products, you can try to disable or uninstall all your browser's extensions/add-ons.
Here are some tips on how to do it in the most common browsers:
Google Chrome
—
Disable your extensions or run the browser in the incognito mode
.
Mozilla Firefox
— Disable your add-ons or run the browser in Save Mode .
Safari
—
Turn all extension off
.
Internet Explorer
—
Disable your add-ons or run the browser in
No Add-ons
mode.
5.1.4 Cannot play voicemails or audio files in Safari
What happens
Do you have any problem with playing voicemails or audio files in Kerio Operator administration or Kerio Phone? Playing of audio files via HTML5 is not possible when the SSL certificate of the hostname is not trusted.
Where it happens
It happens in:
Safari on Apple Mac OS X systems,
Safari on Apple iPhone,
Safari on Apple iPad,
Safari on Windows.
How to fix it
Use a certificate verified by a Certification Authority, or install a self-signed SSL certificate properly and mark it as trusted on your device.
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WARNING
In Safari on Windows you need a certificate verified by a Certification Authority. Self-signed certificates do not work.
How to set self-signed certificate as trusted on Mac OS X systems
1.
Type your Kerio Operator address to the Safari browser. SSL certificate warning appears.
2.
Click
Show Certificate
.
3.
Check
Always trust
.
4.
Click
Continue
.
5.
Enter administrator's username and password for authentication.
How to set self-signed certificate as trusted on Apple iPhone and Apple iPad
1.
Type your Kerio Phone address to browser. The login window appears.
2.
Tap on the
Download SSL certificate
link.
3.
Install the certificate.
NOTE
This certificate will also be used for access to the administration interface too.
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5.2 Vulnerabilities
Vulnerability
Bash vulnerability CVE-
2014-6271,
CVE-2014-7169
(ShellShock)
Linux Glibc vulnerability CVE-
2015-7547
Linux vulnerability CVE-
2015-0235
(GHOST)
OpenSSL vulnerability CVE-
2014-0160
(Heartbleed)
SSL 3.0 vulnerability CVE-
2014-3566 and
POODLE
Description
The shellshock vulnerability (aka CVE-2014-6271 and CVE-2014-7169 ) is a security bug affecting Unix-like operating systems through the Bash shell.
For information on its impact on Kerio products, read Bash vulnerability CVE-2014-6271, CVE-2014-7169
(ShellShock) article.
A vulnerability in the Linux glibc system library has been found. An attacker can gain root access to the server and execute a code.
For more details on its impact on Kerio products, read Linux Glibc vulnerability CVE-2015-7547 article.
There is a vulnerability in Linux glibc system library. An attacker can exploit this vulnerability and gain root access to your server and execute a code.
For more details on its impact on Kerio products, read Linux vulnerability CVE-2015-0235 (GHOST) article.
The National Institute of Standards and Technology (NIST) has published a vulnerability to OpenSSL 1.0.1. Details regarding the vulnerability are available from the NIST website . Kerio Operator 2.2.0 up to 2.2.4 used the affected version of the OpenSSL library. However, a fix is available for Kerio Operator as of version 2.2.5. You can download this release from the Kerio Website .
For additional information and security precautions, read OpenSSL vulnerability CVE-2014-0160 article.
This vulnerability is a flaw in the protocol design. An attacker that controls the network between the client and the server can interfere with any attempted handshake offering TLS 1.0 or later and force both client and server to use SSL 3.0 protocol instead. They can then use other attack techniques (eg. BEAST attack) to decipher transmitted data.
For information on its impact on Kerio products, read SSL 3.0 vulnerability CVE-2014-3566 (POODLE) article.
5.3 USB Tools
This section provides information about password recovery, factory reset, and diagnosing Kerio Operator hardware appliances via an USB flash drive.
5.3.1 Restoring the Kerio Operator default configuration using a USB flash-drive
5.3.2 Restoring the Kerio Operator Box V series default configuration using a USB flash drive
5.3.3 Diagnostic tool for Kerio Operator Box
5.3.4 Diagnostic tool for Kerio Operator Box V series
5.3.5 Recovering password using USB flash-drive for Kerio Operator
5.3.6 Recovering your Kerio Operator Box V series password using a USB flash drive
5.3.1 Restoring the Kerio Operator default configuration using a USB flash-drive
Kerio Technologies provides a set of tools for solutions for situations in which it is not possible to connect to Kerio
Operator on a network and administer it through the Kerio Operator Administration web interface.
These tools are designed to run from a USB flash-drive.
For complete system recovery a USB flash-drive with capacity of at least 1 GB is required. For restoring the default configuration, a capacity of 256 MB is needed.
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Should any issues arise (for example, if Kerio Operator fails to work even after you perform a complete system recovery) please contact our technical support .
Restoring default configuration
The factory settings of Kerio Operator can be recovered with the file kerio-operator-factory-reset
.
Factory settings recovery includes removal of all configuration data including activation and the statistics database.
This USB tools is designed for a single use so that an operation will not repeat if you restart with the flash-drive still in the
USB port. Once you perform the operation, the content cannot be reused, so the file can be removed from the flashdrive.
1.
Insert a USB flash-drive to your computer (256 MB or larger) into a USB port on your computer.
2.
Make sure that only one partition with file system FAT16 or FAT32 (VFAT ) is created on the flash-drive. The USB drive must not be formatted by file system NTFS or ext2, ext3, or ext4.
3.
Save the kerio-operator-factory-reset file to the flash-drive.
4.
Switch off Kerio Operator.
5.
Plug the USB flash-drive into one of the USB ports of your Kerio Operator.
6.
Switch on Kerio Operator.
7.
For factory settings recovery to take effect, Kerio Operator is restarted automatically.
8.
To connect to Kerio Operator, set the following TCP/IP parameters on your computer:
IP address:
10.10.10.2
Subnet mask:
255.255.255.0
9.
Use the web browser of the connected computer to enter the following address: https://10.10.10.1:4021/admin
10.
Set the administrator password, login to the product administration and configure
Kerio Operator Box
as needed.
WARNING
If the steps above do not work, try another flash-drive. Different Kerio Operator Box models require different USB drive formats:
Kerio Operator Box 1210, 3210 and 3230 require a USB flash-drive formatted like a floppy disk (not partitioned).
Kerio Operator Box 1220 requires a flash-drive formatted with a master boot record (MBR). USB drives with floppytype formatting cannot connect to Kerio Operator Box, but can be reconfigured to work. See
Formatting USB flashdrive with MBR
below.
Complete system recovery
Kerio Operator can be completely recovered with the kerio-operator-rescue file. Within the system recovery, all configuration data including activation and the statistics database will be completely rewritten. Therefore the device will have to be reactivated and reconfigured for further use.
Before applying complete system recovery, it is highly recommended to retest connection to Kerio Operator after attempting for
restore of the factory settings .
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Preparing flash-drive for system recovery
For complete system recovery, Kerio Operator Box first needs to introduce operating system from USB drive. File keriooperator-rescue is an image of an installation drive and must be saved directly on the physical device (similarly as in case of burning ISO images on CD). Please follow the instructions according to your client system.
Microsoft Windows
1.
Mount the USB flash-drive to your computer. If necessary, back up files saved on the drive. The flash-drive data will be rewritten completely!
2.
Download and unpack
Image Writer
(it does not require installation).
3.
Download file kerio-operator-rescue .
4.
In application
Image Writer
, look up this file, select your flash-drive and click on
Write
.
5.
Remove the drive securely and unplug it from your computer.
Linux
1.
Mount the USB flash-drive to your computer. If necessary, back up files saved on the drive. The flash-drive data will be rewritten completely!
2.
Download file kerio-operator-rescue
.
3.
Run the terminal (console).
4.
Use command sudo fdisk -l to detect the USB flash-drive name (e.g.
/dev/sdb
).
5.
Save the kerio-operator-rescue file to the USB flash-drive using command: sudo dd if=rescue.img
of=/dev/sdx bs=1M and replace rescue.img
with the real file name and
/dev/diskX with the real appliance. It is necessary to enter the physical device (e.g.
/dev/sdx
), not only a partition (e.g.
/dev/sdx1
).
6.
Use command sudo sync to guarantee finishing of all drive operations.
7.
Unplug the USB drive from your computer.
Mac OS X
1.
Mount the USB flash-drive to your computer. If necessary, back up files saved on the drive. The flash-drive data will be rewritten completely!
2.
Download file kerio-operator-rescue .
3.
Run the terminal (
Applications > Utilities > Terminal
).
4.
Use command sudo diskutil list to detect the USB flash-drive name (e.g.
/dev/DiskX
).
5.
Use command sudo diskutil unmountDisk /dev/diskX to unmount the drive.
6.
Save file kerio-operator-rescue to the USB flash-drive by using command:
sudo dd if=rescue.img of=/dev/Disk1 bs=1m
and replace rescue.img
with the real file name and
/dev/diskX with the real appliance.
7.
Unplug the USB drive from your computer.
Kerio Operator Box device system recovery
1.
Switch off Kerio Operator Box.
2.
Plug the USB flash-drive into one of the USB ports of your
Kerio Operator Box
.
3.
Start the Kerio Operator Box and wait for a sound signal.
4.
To connect to Kerio Operator, set the following TCP/IP parameters on your computer:
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IP address:
10.10.10.2
Subnet mask:
255.255.255.0
5.
Use the web browser of the connected computer to enter the following address: https://10.10.10.1:4021/admin
6.
Set the password, login to the product administration and configure
Kerio Operator Box
as needed.
Recovering USB flash-drive for further use
Special partitions are now created on the USB flash-drive and part of the space is unused. To reuse the drive again as an external desk for other purposes, remove all drive partitions, create one or more new partitions and reformat the drive by an appropriate file system.
Please follow the instructions according to your client system.
Microsoft Windows
1.
Run the
Command Line
.
2.
Enter command diskpart
. On Windows Vista and Windows 7 confirmation of running the application under administration account can be required.
3.
Use command list disk to show the list and look up the number of the physical disk.
4.
Enter command select disk 8
(replace number
8 by the number of the corresponding disk).
5.
Use command clean to remove all created partitions.
6.
Create a new disk partition by using the following commands, as listed: create partition primary select partition 1 format fs=fat32 label="USB Flash" exit
Linux
Use graphical tool GParted or command fdisk
.
Mac OS X
Use system tool Disk Utility (
Application > Utilities > Disk Utility
).
Formatting USB flash-drive with MBR
1.
Connect the USB flash-drive to a computer with Windows operating system.
2.
Run the Command Line.
3.
Enter command diskpart
. On Windows Vista and Windows 7 confirmation of running the application under administration account can be required.
4.
Use command list disk to show the list and look up the number of the physical disk.
5.
Enter command select disk 8
(replace number
8 by the number of the corresponding disk).
6.
Use command clean to remove all created partitions.
7.
Create a new disk partition by using the following commands, as listed:
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create partition primary select partition 1 format fs=fat32 label="USB Flash" quick exit
5.3.2 Restoring the Kerio Operator Box V series default configuration using a USB flash drive
Kerio Technologies provides a set of tools for solutions for situations in which it is not possible to connect to Kerio
Operator on a network and administer it through the Kerio Operator Administration web interface.
You can upgrade the system in your Kerio Operator V series box with a USB flash drive.
You need a flash drive with the capacity of at least 1 GB to run the tools. For restoring the default configuration, 256 MB suffice.
If you have any issues after using the tools, for example, if Kerio Operator fails to work even after you perform a complete system recovery, please contact our technical support
.
Restoring the default configuration
Recovering to factory settings includes removal of all configuration data including activation and the statistics database.
This USB tool is designed for a single use so that the operation does not repeat if you restart with the flash drive still in the USB port. You can remove the files from the flash drive after the upgrade.
1.
Insert the flash drive into a USB port on your computer (256 MB an more).
2.
Make sure that only one partition with the
FAT16
or
FAT32 (VFAT)
file system is created on the flash drive. The USB disk must
not
be formatted by the NTFS, ext2, ext3, or ext4 file systems.
3.
Download and save the kerio-operator-factory-reset file to the flash drive.
4.
Switch off Kerio Operator.
5.
Plug the USB flash drive into a USB port of your Kerio Operator box.
6.
Switch on the Kerio Operator box.
7.
Kerio Operator restarts automatically.
8.
To connect to Kerio Operator, set the following TCP/IP parameters on your computer:
IP address: 10.10.10.2
Subnet mask: 255.255.255.0
9.
In your web browser, type this URL: https://10.10.10.1:4021/admin
10.
Activate the product, login to the product administration interface and configure Kerio Operator as needed.
Running a complete system recovery
During the complete system recovery, all configuration data, including activation and the statistics database, is completely rewritten. This means that you must reactivate and reconfigure the device afterwards.
IMPORTANT
Before doing a complete system recovery,
restore the default configuration
and then retest the connection to
Kerio Operator.
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Preparing a flash drive for a complete system recovery
For a complete system recovery, you must save the installation disk image directly to the physical device.
Microsoft Windows
1.
Insert the flash drive into a USB port on your computer (1 GB an more).
NOTE
All data on the flash drive will be completely overwritten.
2.
Download and unpack
Image Writer
(it does not require installation).
3.
Download the kerio-operator-rescue file.
4.
In Image Writer, find the file, select your flash drive and click
Write
.
5.
Eject the flash drive securely and remove it from your computer.
Linux
1.
Insert the flash drive into a USB port on your computer (1 GB an more).
NOTE
All data on the flash drive will be completely overwritten.
2.
Download the kerio-operator-rescue file.
3.
Run the terminal (console) in the super-user mode (for example, using the su or sudo -s command depending on your Linux distribution).
4.
Use the command fdisk -l to detect the USB flash drive name (for example,
/dev/sdx
).
5.
Save the kerio-operator-rescue file to the flash drive using the following command: dd if=rescue.img
of=/dev/sdx bs=1M and replace rescue.img
with the real file name and
/dev/sdx with the actual device name. You must type the physical device (for example,
/dev/sdx
), not a partition (for example,
/dev/sdx1
).
6.
Use the sync command to ensure all disk operations finish.
7.
Eject the USB drive safely and remove it from the USB port.
Mac OS X
1.
Insert the flash drive into a USB port on your computer (1 GB an more).
NOTE
All data on the flash drive will be completely overwritten.
2.
Download the kerio-operator-rescue file.
3.
Run the terminal:
Applications > Utilities > Terminal
.
4.
Use the sudo diskutil list command to detect the USB flash drive name (for example,
/dev/diskX
).
NOTE
The drive name is case sensitive.
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5.
Use the sudo diskutil unmountDisk /dev/diskX command to unmount the flash drive.
6.
Save the kerio-operator-rescue file to the USB flash drive using the following command: sudo dd if=reescue.img of=/dev/disk1 bs=1m and replace rescue.img
with the real file name and
/dev/diskX with the actual device name.
7.
Eject the flash drive securely and remove it from your computer.
Kerio Operator Box system recovery
1.
Switch off Kerio Operator.
2.
Plug the USB flash drive into a USB port of your Kerio Operator Box.
3.
Switch on Kerio Operator.
4.
To connect to Kerio Operator, set the following TCP/IP parameters on your computer:
IP address: 10.10.10.2
Subnet mask: 255.255.255.0
5.
In your web browser, type this URL: https://10.10.10.1:4021/admin
6.
Activate the product, login to the product administration interface and configure Kerio Operator as needed.
Recovering the USB flash drive for further use
The recovery file creates partitions on the USB flash drive. To reuse the USB drive for other purposes, you need to remove all disk partitions, create new partitions, and reformat the disk for your file system.
Microsoft Windows
1.
Click
Start
and in the
Search
field type cmd.exe
to open the
Command Prompt
window.
2.
In the command line, type diskpart
. You may need to confirm that you have administration rights.
3.
Type list disk to display the list of drives and look up the number of the physical USB drive.
4.
Type select disk X
(replace
X with the number of the corresponding disk).
5.
Type clean to remove all partitions.
6.
Create a new disk partition by typing the following commands in the order listed: create partition primary select partition 1 format fs=fat32 label="USB Flash" quick exit
Linux
Use the graphical tool GParted or the command fdisk
.
Mac OS X
Use the system tool Disk Utility:
Application > Utilities > Disk Utility
.
Formatting a USB flash drive with MBR
1.
Click
Start
and in the
Search
field type cmd.exe
to open the
Command Prompt
window.
2.
In the command line, type diskpart
. You may need to confirm that you have administration rights.
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3.
Type list disk to display the list of drives and look up the number of the physical USB drive.
4.
Type select disk X
(replace
X with the number of the corresponding disk).
5.
Type clean to remove all partitions.
6.
Create a new disk partition by typing the following commands in the order listed: create partition primary select partition 1 format fs=fat32 label="USB Flash" quick exit
Related articles
5.3.3 Diagnostic tool for Kerio Operator Box
Kerio Technologies provides diagnostic tool for diagnose hardware problems with the Kerio Operator Box, which elicits crucial information for the Kerio Technologies technical support. The tool is designed for use from a USB flashdisk.
You need a USB flasdisk with capacity of at least 256 MB.
The diagnostics tool is designed for a single use to avoid unexpected repetition of the operation upon the next restart in case that the flashdisk has not been dismounted. This implies that once you perform the operation, the disk content cannot be used again and the files can be removed.
Using a diagnostic USB tool
NOTE
Kerio Operator uses the same diagnostic tool as Kerio Control. For diagnostics, use the kerio-controlusbdiag file.
If you need the USB diagnostic tool, download and use the diagnostic tool
.
Creating diagnostic flashdisk
The diagnostic tool is an image of an installation disk and must be saved directly on the physical device (similarly as in case of burning ISO images on CD). Please follow the instructions according to your client system.
Microsoft Windows
1.
Mount the USB flashdisk to your computer. If necessary, back up files saved on the disk. The flashdisk data will be rewritten completely!
2.
Download and unpack Image Writer (it does not require installation).
3.
Download file the diagnostic tool file.
4.
In application
Image Writer
, look up this file, select your flashdisk and click on
Write
.
5.
Remove the disk securely and unplug it from your computer.
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Linux
1.
Mount the USB flashdisk to your computer. If necessary, back up files saved on the disk. The flashdisk data will be rewritten completely!
2.
Download the diagnostic tool file.
3.
Run the terminal (console).
4.
Use command sudo fdisk -l to detect the USB flashdisk name (e.g.
/dev/sdb
).
5.
Save the diagnostic tool file on this device by using the following command: sudo dd if=usbdiag.img of=/dev/sdx bs=1M
Replace string with: usbdiag.img
by the real file name and
/dev/sdx by the real device. It is necessary to enter the physical device
(e.g.
/dev/sdx
), not only a fragment (for example,
/dev/sdx1
).
6.
Use command sudo sync to guarantee finishing of all disk operations.
7.
Unplug the USB disk from your computer.
Mac OS X
1.
Mount the USB flashdisk to your computer. If necessary, back up files saved on the disk. The flashdisk data will be rewritten completely!
2.
Download the diagnostic tool file.
3.
Run the terminal (
Applications > Utilities > Terminal
).
4.
Use command sudo diskutil list to detect the USB flashdisk name (e.g.
/dev/diskX or
/dev/DiskY
— watch the letter case).
5.
Use command sudo diskutil unmountDisk /dev/diskX to unmount the disk.
6.
Save the diagnostic tool file on the USB disk by using the following command: sudo dd if=usbdiag.img of=/dev/disk1 bs=1m
Replace string with: usbdiag.img
by the real file name and
/dev/diskX by the real device.
7.
Unplug the USB disk from your computer.
Using diagnostic flashdisk
1.
Switch off Kerio Operator Box.
2.
Plug the USB flashdisk into one of the USB ports of your Kerio Operator Box.
3.
Switch on Kerio Operator Box.
4.
Approximately after two minutes Kerio Operator Box beeps three times. This means that the operating system has been introduced and the diagnostic test has just been started. If the device does not beep within the following 10 minutes, the test has failed. In such case switch off the device, unplug the USB flashdisk and send diagnostic information the
Kerio Technologies technical support (see below).
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5.
The test starts with 10 beeps and runs for about 60 minutes — a 40-minute memory test and a diagnostic test. If you want to skip the memory test, press any key during the ten-beep interval. Once the test is finished, Kerio Control Box starts beeping every 30 seconds.
6.
Switch off Kerio Operator Box and unplug the USB flashdisk.
Test results processing
Plug the USB flashdisk to your computer again. There is a partition called
KerioDiag on the disk. This partition includes the file with test results.
Please send this file to the Kerio Technologies technical support and possibly provide a description of the non-standard behavior of your Kerio Operator Box.
Recovering USB flashdisk for further use
For more information, refer to Recovering USB flash-drive for further use (page 313).
5.3.4 Diagnostic tool for Kerio Operator Box V series
Kerio Technologies provides a tool for diagnosing hardware problems with the Kerio Operator Box NG series. This tool collects crucial information for the Kerio Technologies technical support. It is designed to be run from a USB flash drive.
You need a USB flash drive with a capacity of at least 256 MB.
The diagnostic tool is designed for a single use so that the operation will not repeat if you restart with the flash drive still in the USB port. Once you perform the operation, the content cannot be reused, so the files can be removed from the flash drive.
Creating the diagnostic flash drive
NOTE
Kerio Operator uses the same diagnostic tool as Kerio Control. For diagnostics, use the kerio-controlusbdiag file.
The diagnostic tool is an image of an installation disk and must be saved directly on the physical device. Follow the instructions for your client system below.
Microsoft Windows
1.
Insert the flash drive to your computer (256 MB or larger) into a USB port on your computer.
NOTE
All data on the flash drive will be completely overwritten, so be sure to save any files you need elsewhere.
2.
Download and unpack
Image Writer
(it does not require installation).
3.
Download the diagnostic tool file.
4.
In
Image Writer
, find the file, select your flash drive and click
Write
.
5.
Eject the flash drive securely and remove it from your computer.
Linux
1.
Insert the flash drive into a USB port on your computer.
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NOTE
All data on the flash drive will be completely overwritten, so be sure to save any files you need elsewhere.
2.
Download the diagnostic tool file.
3.
Run the terminal (console).
4.
Use the command sudo fdisk -l to detect the USB flash drive name (e.g.,
/dev/sdx
).
5.
Save the diagnostic tool file on the flash drive using this command: sudo dd if=usbdiag.img
of=/dev/sdx bs=1M
(replace usbdiag.img
with the real file name and
/dev/sdx with the actual device name). You must enter the physical device (e.g.,
/dev/sdx
), not the partition (e.g.,
/dev/sdx1
).
6.
Use the command sudo sync to ensure that all disk operations finish.
7.
Eject the flash drive securely and remove it from your computer.
Mac OS X
1.
Insert the flash drive into a USB port on your computer.
NOTE
All data on the flash drive will be completely overwritten, so be sure to save any files you need elsewhere.
2.
Download the diagnostic tool file.
3.
Run the terminal:
Applications > Utilities > Terminal
.
4.
Use the command sudo diskutil list to detect the USB flash drive name (e.g.,
/dev/diskX or
/dev/DiskY
).
NOTE
This is case sensitive.
5.
Use the command sudo diskutil unmountDisk /dev/diskX to eject the flash drive.
6.
Save the diagnostic tool file on the USB flash drive using this command: sudo dd if=usbdiag.img
of=/dev/disk1 bs=1m
(Replace string usbdiag.img
with the real file name and
/dev/diskX with the real device).
7.
Eject the flash drive securely and remove it from your computer.
Using the diagnostic flash drive
1.
Switch off Kerio Operator Box.
2.
Plug the USB flash drive into one of the USB ports of your
Kerio Operator Box
.
3.
Switch on Kerio Operator Box to run the diagnostic test. The diagnostic test may take some time (approximately 60 minutes).
4.
Switch off Kerio Operator Box and eject the USB flash drive.
Test results processing
Reinsert the flash drive into the USB port.
Find the partition called
KerioDiag
on the flash drive. It contains the file with test results.
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Send this file to Kerio Technologies technical support, and optionally provide a description of the nonstandard behavior of your Kerio Operator Box.
Recovering USB flash drive for further use
To reuse your flash drive, you will need to reformat it to remove the partitions.
Related articles
5.3.5 Recovering password using USB flash-drive for Kerio Operator
Kerio Technologies provides a tool for password recovery. The tool is designed for use from a USB flash-drive.
For password recovery a USB flash-drive with capacity of at least 256 MB will do.
The password recovery tool is designed for a single use to avoid unexpected repetition of the operation upon the next restart in case that the flash-drive has not been dismounted. This implies that once you perform the operation, the disk content cannot be used again and the files can be removed.
Creating and using a password recovery tool
Forgotten administration password can be recovered by using file kerio-operator-password-reset .
Follow these instructions:
1.
Mount the USB flash-drive to your computer.
2.
Make sure that only one fragment with file system
FAT16
or
FAT32
(
VFAT
) is created on the flash-drive. The USB disk must not be formatted by file system
NTFS
or
ext2 / ext3 / ext4
.
3.
Save file kerio-operator-password-reset to the flash-drive.
4.
Switch off the Kerio Operator.
5.
Plug the USB flash-drive into the USB port.
6.
Switch on Kerio Operator.
7.
In your web browser, open the Kerio Operator Administration.
8.
Activation wizard opens in the browser. As the product has already been activated, the wizard will require a new administration password.
Now you can login as user
Admin with a new password.
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WARNING
If it does not work, try another USB flash-drive.
Details of the known issue: There are two formats of USB flash-drives (the first type uses MBR as a boot sector, the second one is formatted as a floppy). Each format is eligible for different type of hardware device:
Kerio Operator Box 1210, 3210 and 3230 — if the USB flash-drive is formatted as a floppy. The second type which uses MBR cannot connect to Kerio Operator Box.
Kerio Operator Box 1220 — if the USB flash-drive uses MBR as a boot record, you can use it directly. The second type which is formatted as a floppy cannot connect to Kerio Operator Box. If you want to use it, format the USB flashdrive according to the following steps:
Formatting USB flash-drive with MBR
5.3.6 Recovering your Kerio Operator Box V series password using a USB flash drive
You can recover your password for the administration interface with a USB flash drive.
You need a flash drive with the capacity of at least 256 MB.
The password recovery tool is designed for a single use so that the operation does not repeat if you restart with the flash drive still in the USB port. You can remove the files from the flash drive after the upgrade.
Creating and using a password recovery tool
To recover a lost administration password:
1.
Insert the flash drive into a USB port on your computer.
2.
Make sure that only one partition with the
FAT16
or
FAT32 (VFAT)
file system is created on the flash drive. The USB disk must
not
be formatted by the NTFS, ext2, ext3, or ext4 file systems.
3.
Download and save the file kerio-operator-password-reset to the flash drive.
4.
Switch off Kerio Operator.
5.
Plug the USB flash drive into a USB port of your Kerio Operator box.
6.
Switch on Kerio Operator.
7.
When the Kerio Operator Engine starts up, open the Kerio Operator administration interface in a browser. The activation wizard opens.
8.
In the activation wizard, create a new password for the admin account.
Related articles
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6 Glossary
A
Analog telephone adapter
A device for connecting analog devices to a digital or voice over IP network.
ATA
Analog telephone adapter - A device for connecting analog devices to a digital or voice over IP network.
B
Basic Rate Interface
An ISDN channel intended for small systems which achieves up to 128kbps data rate.
BLF
Busy Lamp Field - A set of indicators that monitor the current state (online, offline, busy, on a call) of a phone extension.
BRI
Basic Rate Interface - An ISDN channel meant for small enterprise systems to obtain upto 128kbps data rate.
Busy Lamp Field
A set of indicators that monitor the current state (online, offline, busy, on a call) of a phone extension.
C
Cacti
Monitoring tool based on SNMP.
call routing
A process for routing of incoming and outgoing calls between internal extensions, PBX, and for example
PSTN.
Caller ID
A service that provides information about caller's number.
Click to dial
An action that requests a real time connection via phone call.
Codecs
Programs used in streaming media and audio/video conferencing that encodes or decodes digital data streams and signals.
CPE
Customer-premises equipment - Provider's devices that are physically located on the customer's premises.
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CRM
Customer Relationship Management - Strategies and technologies for managing and analyzing customer relationships.
Customer-premises equipment
Provider's devices that are physically located on the customer's premises and are connected to a telecommunication channel of the provider.
Customer Relationship Management
Strategies and technologies for managing and analyzing customer relationships.
D
DECT
Digital Enhanced Cordless Telecommunications - Cordless telephone systems.
DHCP
Dynamic Host Configuration Protocol - A protocol that automatically gives IP addresses and additional configuration to hosts in a network.
Digital Enhanced Cordless Telecommunications
Cordless telephone systems.
DTMF
Dual-tone multi-frequency signal - Tone generated by the telephone or fax device when dialling while communicating with the telephone line provider.
Dual Tone - Multi Frequency
The tone generated by the telephone (or fax) device when dialling. This is used for communicating with the telephone line provider.
Dynamic Host Configuration Protocol
A protocol that automatically gives IP addresses and additional configuration to hosts in a network.
E
Euro-ISDN
An Integrated Services Digital Network (ISDN) standards as developed by European Telecommunications
Standards Institute (ETSI).
F
File Transfer Protocol
A protocol for tranferring computer files from a server.
FTP
File Transfer Protocol - A protocol for tranfering computer files from a server.
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H hardware appliance
Kerio Operator installed and delivered with standardized and tested hardware box.
I
Internet telephony service provider
A type of a service provider that provides communication via Internet. The communication is based on
Voice over Internet Protocol (VoIP).
IP PBX
A Private Branch Exchange system that connects telephone extensions through internet and provides additional audio and video communication features.
ISDN
Integrated services digital network - A technology enabling digital transmission of data and voice signals over a telephone network.
ITSP
Internet telephony service provider - A type of a service provider that provides communication based on
Voice over Internet Protocol (VoIP).
K
Kerio Operator App for Salesforce
An application based on Call Center that integrates Kerio Operator and Salesforce.
Kerio Operator Softphone
A softphone app for Android or iOS.
Kerio Phone
A softphone available as a native desktop or web-based application.
L
LDAP
Lightweight Directory Access Protocol enables users to access centrally managed contacts.
Lightweight Directory Access Protocol
Lightweight Directory Access Protocol enables users to access centrally managed contacts.
M
Master Boot Record
A type of a boot sector.
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MBR
Master Boot Record - A type of a boot sector.
MyKerio
A web-based application for monitoring and managing appliances of Kerio products.
N
NAT
Network address translation - A method that remaps IP addresses by changing network address information.
Network address translation
A method that remaps IP addresses by changing network address information.
P
PBX
Private Branch Exchange - System that connects telephone extensions and switches calls.
prefix
Country codes, area codes, a number, or a set of numbers that are dialed before the telephone numbers.
PRI
Primary Rate Interface - An Integrated Services Digital Network channel for large enterprise systems to obtain higher speed than Basic Rate Interface.
Primary Rate Interface
An ISDN channel meant for large enterprise systems to obtain higher data transfer rate than Basic Rate
Interface.
PSTN
Public switched telephone network - A global telecommunications network that operates the traditional telephony system.
Public switched telephone network
A global telecommunications network that operates the traditional telephony system.
Q
QoS
Quality of service - Network's ability to obtain maximum bandwidth and manage other network performance elements like latency, error rate and uptime.
S
Secure Sockets Layer
A protocol that ensures integral and secure communication between networks.
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Service record
Service record is a record in DNS that specifies the location of server for individual services.
Session Initiation Protocol
A communication protocol used for voice and video calls in Internet telephony or private IP telephone systems.
Simple Network Management Protocol
A protocol for gathering and organizing information about devices in IP networks, and changing devices behavior.
SIP
Session Initiation Protocol - A communication protocol used for voice and video calls in Internet telephony or private IP telephone systems.
SIP interface
An external interface used for connecting to SIP providers.
SIP password
A password for authenticating provided by a SIP provider.
SIP Provider
A telecommunications company that provide telephony services based on Voice over Internet Protocol.
SIP trunk
Wide range of external numbers provided by a SIP provider.
SIP username
An username for authenticating provided by a SIP provider.
SMTP
Simple Mail Transport Protocol - An internet standard used for email transmission across IP networks.
SNMP
Simple Network Management Protocol - A protocol for gathering and organizing information about devices in IP networks, and changing devices behavior.
Software Appliance
A special operating system designed to be installed on a computer.
SRVrecord
Service record is a record in DNS that specifies the location of server for individual services.
SSL
Secure Sockets Layer - A protocol that ensures integral and secure communication between networks.
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T
Telecommunications service provider
A type of a service provider that provides telephone and other services.
TFTP
Trivial File Transfer Protocol - A simple protocol for transferring files.
Trivial File Transfer Protocol
A simple protocol for transferring files.
TSP
Telecommunications service provider - A type of a service provider that provides telephone and other services.
U
UDP
User Datagram Protocol - Ensures packet transmission.
URL
The Uniform Resource Locator is the address of a web page on the world wide web.
User Datagram Protocol
Ensures packet transmission.
V
Virtual Appliance
Pre-configured Kerio Operator virtual machine image for VMware.
Voice over Internet protocol
A digital telephone system that uses the internet as the transmission medium, rather than the PSTN.
VoIP
Voice over Internet protocol - A digital telephone system that uses the internet as the transmission medium, rather than the PSTN.
VoIP Phone
A device that works on principles of VoIP.
W webRTC
Set of communication protocols enabling real-time communication over peer-to-peer connections.
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7 Legal Notices
7.1 Trademarks and registered trademarks
Aastra
® is registered trademark of Aastra Technologies Limited.
Active Directory
® is registered trademark of Microsoft Corporation.
Cisco
® and Linksys
® are registered trademarks of Cisco Systems, Inc.
Digium
® is registered trademark of Digium, Inc.
Firefox
® is registered trademark of Mozilla Foundation.
Grandstream
® is a registered trademark of Grandstream Networks, Inc.
Internet Explorer
® is registered trademark of Microsoft Corporation.
Polycom
® is registered trademark of Polycom, Inc.
Safari™ is registered trademark of Apple Inc.
Salesforce
® and Salesforce.com
® are registered trademarks of salesforce.com, Inc.
SJphone
® is registered trademark of SJ Labs, Inc.
snom
® is registered trademark of snom technology AG.
snom
® is registered trademark of snom technology AG.
Wireshark
® is registered trademark of Wireshark Foundation.
X-Lite is a software phone developed by CounterPath Corporation with registered trademark of CounterPath
®
.
7.2 Used open source software
This product contains the following open-source libraries:
adapter.js
Shim to insulate apps from spec changes and prefix differences.
Copyright (c) 2014, The WebRTC project authors. All rights reserved.
Appliance OS Sources
Kerio Operator devices are based on open software from various resources. For detailed information on conditions of each particular software used in the product, refer to acknowledgments.
To download the source package, go to http://download.kerio.com/archive/
.
asterisk
Asterisk - An open source telephony toolkit.
Copyright © 1999 - 2012 Digium, Inc. and others.
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AudioContext-Polyfill
Polyfill for AudioContext and its parties on Web Audio API.
Copyright © 2013 - 2014 Shinnosuke Watanabe
coturn
coturn TURN server project
Copyright © 2011, 2012, 2013 Citrix Systems
Heimdal Kerberos
Heimdal is an implementation of Kerberos 5, largely written in Sweden. It is freely available under a three clause BSD style license (but note that the tar balls include parts of Eric Young's libdes, which has a different license). Other free implementations include the one from MIT, and Shishi. Also Microsoft Windows and Sun's Java come with implementations of Kerberos.
Copyright © 1997-2000 Kungliga Tekniska Hogskolan (Royal Institute of Technology, Stockholm, Sweden). All rights reserved.
Copyright © 1995-1997 Eric Young. All rights reserved.
Copyright © 1990 by the Massachusetts Institute of Technology
Copyright © 1988, 1990, 1993 The Regents of the University of California. All rights reserved.
Copyright © 1992 Simmule Turner and Rich Salz. All rights reserved.
jsonrpccpp
C++ framework for json-rpc (json remote procedure call)
Copyright © 2011-2014 Peter Spiess-Knafl
Kerio Asterisk Module
The Kerio Asterisk Module extends the functionality of the Asterisk PBX to match Kerio Operator needs. It is distributed and licensed under GNU General Public License version 2. The complete source code is available at: http://download.kerio.com/archive/
Copyright © 2010 Kerio Technologies s.r.o
© Copyright 2000-2006 T.I.P Group S.A. and the IBPP Team (www.ibpp.org).
libcurl
Libcurl is a free and easy-to-use client-side URL transfer library. This library supports the following protocols: FTP, FTPS,
HTTP, HTTPS, GOPHER, TELNET, DICT, FILE and LDAP.
Copyright © 1996-2008, Daniel Stenberg.
libiconv
Libiconv converts from one character encoding to another through Unicode conversion.
Copyright © 1999-2003 Free Software Foundation, Inc.
Author: Bruno Haible
Homepage: http://www.gnu.org/software/libiconv/
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The libiconv library is distributed and licensed under GNU Lesser General Public License version 3.
Kerio Operator includes a customized version of this library. Complete source codes of the customized version of libiconv library are available at: http://download.kerio.com/archive/
libmbfl
libmbfl is a streamable multibyte character code filter and converter library. The libmbfl library is distributed under LGPL license version 2.
Copyright © 1998-2002 HappySize, Inc. All rights reserved.
The library is available for download at: http://download.kerio.com/archive/
libopus
Opus is a high-quality audio codec developed in cooperation among Xiph.org, Broadcom, and Microsoft (Skype). The codec is standardized in RFC 6716. The reference implementation of the codec is licensed under a 3-clause BSD-style license. The copyright and patent licenses for the Opus algorithm are automatically granted to everyone and do not require application or approval. The patent licenses are included below together with the BSD-style license.
Copyright © 2011-2014 Opus contributors
libxml2
XML parser and toolkit.
Copyright © 1998-2003 Daniel Veillard. All Rights Reserved.
Copyright © 2000 Bjorn Reese and Daniel Veillard.
Copyright © 2000 Gary Pennington and Daniel Veillard
Copyright © 1998 Bjorn Reese and Daniel Stenberg.
nginx-nchan
Fast, horizontally scalable, multiprocess pub/sub queuing server and proxy for HTTP, long-polling, Websockets and
EventSource (SSE), powered by Nginx.
https://nchan.slact.net/
Written by Leo Ponomarev (slact) 2009-2015.
nginx-upload-module
A module for nginx web server for handling file uploads using multipart/form-data encoding (RFC 1867).
http://www.grid.net.ru/nginx/upload.en.html
Copyright © 2006, 2008, Valery Kholodkov
OpenLDAP
Freely distributable LDAP (Lightweight Directory Access Protocol) implementation.
Copyright © 1998-2007 The OpenLDAP Foundation
Copyright © 1999, Juan C. Gomez, All rights reserved
Copyright © 2001 Computing Research Labs, New Mexico State University
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Portions Copyright © 1999, 2000 Novell, Inc. All Rights Reserved
Portions Copyright © PADL Software Pty Ltd. 1999
Portions Copyright © 1990, 1991, 1993, 1994, 1995, 1996 Regents of the University of Michigan
Portions Copyright © The Internet Society (1997)
Portions Copyright © 1998-2003 Kurt D. Zeilenga
Portions Copyright © 1998 A. Hartgers
Portions Copyright © 1999 Lars Uffmann
Portions Copyright © 2003 IBM Corporation
Portions Copyright © 2004 Hewlett-Packard Company
Portions Copyright © 2004 Howard Chu, Symas Corp.
OpenSSL
An implementation of Secure Sockets Layer (SSL v2/v3) and Transport Layer Security (TLS v1) protocol.
This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit
( http://www.openssl.org/ ).
This product includes cryptographic software written by Eric Young.
This product includes cryptographic software written by Tim Hudson.
PHP
PHP is a widely-used scripting language that is especially suited for Web development and can be embedded into
HTML.
Copyright © 1999-2006 The PHP Group. All rights reserved.
This product includes PHP software, freely available from http://www.php.net/software/
php-ev
ev provides interface to libev library - high performance full-featured event loop written in C.
Copyright © 2012,2013,2014 Ruslan Osmanov <[email protected]>
PHP-JWT
A simple library to encode and decode JSON Web Tokens (JWT) in PHP, conforming to RFC 7519.
Copyright © 2011, Neuman Vong
pjproject
Asterisk fork of PJSIP
Copyright © 2003-2008 Benny Prijono <[email protected]>
Copyright © 2008-2011 Teluu Inc. ( http://www.teluu.com
)
ScoopyNG
This product includes software developed by Tobias Klein.
Copyright © 2008, Tobias Klein. All rights reserved.
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SIP.js
A simple, intuitive, and powerful JavaScript signaling library http://sipjs.com
Copyright © 2014 Junction Networks, Inc.
http://www.onsip.com
tftpd
TFTP daemon. TFTP is a simple protocol used for file transmission.
Copyright © 1983 Regents of the University of California. All rights reserved.
uwsgi
uWSGI application server container http://projects.unbit.it/uwsgi
Copyright © 2009-2014 Unbit S.a.s. <[email protected]>
WAVPlayerProject
WAV player.
Denis Kolyako May 28, 2007, see http://etcs.ru/copyright/
zlib
General-purpose library for data compressing and decompressing.
Copyright © 1995-2005 Jean-Loup Gailly and Mark Adler.
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Key Features
- Enterprise-class voice and video communication
- Easy to administer & flexible to deploy
- Support for high quality codecs
- Automatic provisioning feature
- Advanced security technologies
- Call handling features
- Centralized web interface
Frequently Answers and Questions
What is Kerio Operator?
How to connect Kerio Operator to network?
How to login to Kerio Operator Administration?
Related manuals
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