Administration Manual

Transcription

Series: 3.12
Rev.: 08-05-2015
Web: www.kalliopepbx.it
Technical support: http://ticket.netresults.it - [email protected]
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Rev.: 08-05-2015
Correct disposal of batteries and electronic devices
Batteries composition: Lithium
(Applicable in countries with separate collection systems)
This marking on the battery, manual or packaging indicates that the
batteries in this product should not be disposed of with other
household waste. Where marked, the chemical symbols Hg, Cd or Pb
indicate that the battery contains mercury, cadmium or lead above
the reference levels in EC Directive 2006/66. If batteries are not
properly disposed of, these substances can cause harm to human
health or the environment. To protect natural resources and to
promote material reuse, please separate batteries from other type of
waste and recycle them through your local, free battery return
system.
Please contact your local County Office or the device reseller for further info on how to
properly dispose batteries and the device itself.
Support
In case of malfunctioning please contact your reseller.
NetResults S.r.l.
Via Agostino Bassi, 5 – 56121 Ospedaletto (PI)
Tel. +39 050 3163658
Web: http://www.netresults.it – E-mail: [email protected]
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Index
Chapter 1 System Management .......................................................................8
1.1
Introduction .......................................................................................8
1.2
Administration GUI ...............................................................................9
1.3
A General Overview ............................................................................ 10
1.3.1 Phone calls originated from a PBX extension .......................................... 11
1.3.2 Incoming calls ................................................................................ 12
Chapter 2 System Menu .............................................................................. 14
2.1
Status ............................................................................................. 14
2.1.1 Firmware version ............................................................................ 14
2.1.2 Date and time................................................................................ 14
2.1.3 VoIP Services ................................................................................. 15
2.1.4 SIP PnP Service .............................................................................. 15
2.2
Network .......................................................................................... 16
2.3
Network Services ............................................................................... 17
2.3.1 NAT Traversal via external IP ............................................................. 18
2.4
SNMP .............................................................................................. 18
2.5
Firmware Update ............................................................................... 19
2.5.1 Update Process .............................................................................. 19
2.6
AMI ................................................................................................ 21
2.7
Configuration Backup .......................................................................... 21
2.8
KCTI/Web GUI Users ........................................................................... 22
2.9
Administrator Settings ......................................................................... 24
2.10
High Availability ................................................................................ 24
2.10.1
Enabling HA ............................................................................... 25
2.10.2
Disabling HA .............................................................................. 28
2.11
Licenses .......................................................................................... 29
2.11.1
G.729 Licenses ........................................................................... 29
2.11.2
Kalliope Licenses ......................................................................... 31
Chapter 3 PBX Menu .................................................................................. 32
3.1
Extensions ....................................................................................... 32
3.1.1 Global Settings .............................................................................. 32
3.1.2 Local Extensions ............................................................................. 34
3.1.3 Add or modify a local extension .......................................................... 35
3.1.4 Importing custom lists of extensions from CSV/XLS files ............................ 41
3.1.5 Remote extensions .......................................................................... 42
3.2
Call Groups and Automatic Call Distribuition.............................................. 43
3.2.1 Call Groups ................................................................................... 44
3.2.2 Automatic Call Distribuition ............................................................... 45
3.3
Gateway .......................................................................................... 48
3.3.1 Options ........................................................................................ 48
3.3.2 Gateways/VoIP Terminations and Trunks ............................................... 50
3.4
IVR................................................................................................. 61
3.5
Features .......................................................................................... 62
3.5.1 On-call Services ............................................................................. 63
3.5.2 SIP Settings ................................................................................... 64
3.5.3 Voice Mailbox ................................................................................ 64
3.5.4 Audio Codec Selection...................................................................... 64
3.6
Incoming Calls Policy .......................................................................... 65
3.6.1 Opening Time Check ........................................................................ 65
3.6.2 Incoming Calls Policy ....................................................................... 67
3.6.3 Callback Service ............................................................................. 68
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3.6.4 DISA Service .................................................................................. 69
3.7
Numbering Plan ................................................................................. 69
3.7.1 Call-pickup Group ........................................................................... 70
3.7.2 Direct Pickup ................................................................................. 71
3.7.3 Call Pickup with Invitation ................................................................ 71
3.7.4 Callback Request ............................................................................ 71
3.7.5 Echo Service.................................................................................. 71
3.7.6 Voice Mailbox ................................................................................ 71
3.7.7 Audioconference Service .................................................................. 71
3.7.8 DISA Service .................................................................................. 71
3.7.9 Electronic Lock .............................................................................. 71
3.7.10
Unconditional Forward .................................................................. 71
3.7.11
Call Tagging (available with the optional Call Center module) ................. 72
3.7.12
Spy call supervision ...................................................................... 72
3.7.13
Dual Ring .................................................................................. 72
3.7.14
Call Parking ............................................................................... 72
3.8
Outbound Routing .............................................................................. 74
Chapter 4 PBX Applications ......................................................................... 76
4.1
External API ..................................................................................... 76
4.2
MeetMe Conference ............................................................................ 78
Chapter 5 Phonebook ................................................................................ 80
5.1
Phonebook ....................................................................................... 80
5.1.1 ‘Admin’ login................................................................................. 80
5.1.2 ‘User’ login ................................................................................... 83
5.2
Publish extensions on LDAP ................................................................... 85
5.3
Publish Shared Phonebook on LDAP ......................................................... 86
5.4
Import from LDAP .............................................................................. 86
Chapter 6 CDR ......................................................................................... 88
6.1
Call Detail Record .............................................................................. 88
Chapter 7 -
Call Recording ............................................................................ 90
Chapter 8 -
Troubleshooting .......................................................................... 91
Chapter 9 Provisioning ............................................................................... 92
9.1
Auto-provisioning procedure ................................................................. 92
9.2
Provisioning procedure steps ................................................................. 93
9.2.1 Extensions creation ......................................................................... 93
9.2.2 Devices configuration ...................................................................... 93
9.2.3 Configuration Template definition ....................................................... 94
9.2.4 Configuration file creation ................................................................ 95
9.2.5 Yealink phones templates ................................................................. 97
9.2.6 Linksys phones templates .................................................................. 97
9.2.7 Snom phone templates ..................................................................... 98
Chapter 10 File Manager .............................................................................. 99
10.1
Audio Files ....................................................................................... 99
Appendix A: Provisioning .................................................................................. 101
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1.1 Login screen .......................................................................................8
1.2 KalliopePBX configuration interface ..........................................................9
2.1 System  Status ................................................................................ 14
2.2 Manual date and time setting ................................................................ 15
2.3 System  Network ............................................................................. 16
2.4 System  Network Services .................................................................. 17
2.5 System  NAT Traversal ...................................................................... 18
2.6 System  SNMP ................................................................................. 18
2.7 System  Update .............................................................................. 19
2.8 'Reboot with secondary firmware' button .................................................. 20
2.9 System  AMI ................................................................................... 21
2.10 System  Backup ............................................................................. 21
2.11 Restoring configuration backup ............................................................ 22
2.12 Alert of uncomplete restore process ...................................................... 22
2.13 System Web GUI Users ..................................................................... 23
2.14 System  User access credentials screen ................................................ 23
2.15 System  Settings ............................................................................ 24
2.16 HA configuration screen ..................................................................... 25
2.17 HA status screen, acquisition in process .................................................. 26
2.18 HA status screen, acquisition complete .................................................. 26
2.19 HA status screen, pairing is not possible ................................................. 27
2.20 HA status screen, HA is enabled on South node ......................................... 27
2.21 HA status screen, ongoing pairing ......................................................... 27
2.22 HA North node status screen, pairing process complete ............................... 28
2.23 HA South node screen, pairing complete ................................................. 28
2.24 System  Licenses............................................................................ 29
2.25 G.729 License Key Activation screen (Digium License) ................................ 30
2.26 G.729 License Key Activation screen (Personal Info) ................................... 30
3.1 PBX  Extensions .............................................................................. 32
3.2 PBX  Extensions  Global Settings ....................................................... 32
3.3 Extensions unlock screen...................................................................... 33
3.4 Failover actions configuration screen ...................................................... 34
3.5 Extension configuration screen, Account .................................................. 35
3.6 Extension configuration screen, Identity ................................................... 36
3.7 Extension configuration screen, Publication .............................................. 37
3.8 Extension configuration screen, Call-Pickup Group ...................................... 37
3.9 Extension configuration screen, Outbound calls .......................................... 38
3.10 Extension configuration screen, Voice mailbox ......................................... 38
3.11 Extension configuration screen, Failover actions ....................................... 39
3.12 Extension configuration screen, Busy Lamp Field ....................................... 39
3.13 Codec Selection ............................................................................... 40
3.14 Extension configuration screen, Call Recording ......................................... 40
3.15 Extensions import process ................................................................... 41
3.16 Extensions recapitulation screen ........................................................... 42
3.17 List of remote extensions.................................................................... 42
3.18 Remote extensions editing screen ......................................................... 42
3.19 PBX  Call Groups ............................................................................ 43
3.20 Call Group editing screen ................................................................... 44
3.21 Call Group Time Check ....................................................................... 45
3.22 Enabling Call Recording for Call Groups .................................................. 45
3.23 ACD editing screen ............................................................................ 46
3.24 ACD editing, Members ........................................................................ 46
3.25 ACD editing, Callers .......................................................................... 47
3.26 Global options for outbound calls .......................................................... 48
3.27 Gatewats/VoIP Terminations and Trunks ................................................. 50
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Figure 3.28 Add/Editing Gateway screen ................................................................ 52
Figure 3.29 PBX  Gateways  Add/Edit Gateway, DID configuration screen .................. 54
Figure 3.30 PBX  Gateway  Add new VoIP termination, VoIP domain selection ............. 55
Figure 3.31 PBX  Gateway  Add new VoIP termination, Account .............................. 56
Figure 3.32 PBX  Gateway  Add new VoIP Termination, Configuration ....................... 56
Figure 3.33 PBX  Gateway  Add new VoIP Termination, Outbound Calls Policy ............. 56
Figure 3.34 PBX  Gateway  Add new VoIP Termination, DID configuration screen ......... 57
Figure 3.35 VoIP Domain summary screen............................................................... 57
Figure 3.36 PBX  Gateway  Add new Trunk, Outbound Calls Policy ........................... 59
Figure 3.37 PBX  Gateway  Add new Trunk, Remote numbering range ....................... 59
Figure 3.38 PBX  Gateway  Add New Balancer .................................................... 60
Figure 3.39 PBX  IVR ...................................................................................... 61
Figure 3.40 IVR Configuration ............................................................................. 61
Figure 3.41 PBX  Features ............................................................................... 63
Figure 3.42 PBX  Incoming Calls Policy ................................................................ 65
Figure 3.43 Time Check configuration screen .......................................................... 66
Figure 3.44 Closing days configuration screen .......................................................... 66
Figure 3.45 Opening time configuration screen ........................................................ 67
Figure 3.46 Callback service configuration .............................................................. 68
Figure 3.47 Callback enabled numbers configuration screen ........................................ 68
Figure 3.48 DISA Service configuration screen ......................................................... 69
Figure 3.49 PBX  Internal Numbering Plan ............................................................ 70
Figure 3.50 PBX  Internal Numbering Plan, Services configuration screen ..................... 70
Figure 3.51 Internal Numbering Plan, Custom Numbering Plan rules .............................. 72
Figure 3.52 Custom Numbering Plan rules configuration screen .................................... 73
Figure 3.53 PBX  Numbering Plan, Local and Remote Extensions and Outbound Routing .... 73
Figure 3.54 PBX  Outbound routing .................................................................... 74
Figure 4.1 New external API configuration, Global settings ......................................... 76
Figure 4.2 New external API configuration, Parameters acquisition ............................... 77
Figure 4.3 New external API configuration, Actions configuration ................................. 78
Figure 5.1 Phonebook ....................................................................................... 80
Figure 5.2 Shared Phonebook section .................................................................... 81
Figure 5.3 Add new contact to Personal/Shared Phonebook screen ................................ 82
Figure 5.4 Import from CSV/XLS screen ................................................................. 82
Figure 5.5 Contacts Import preview ...................................................................... 83
Figure 5.6 Personal Phonebook section .................................................................. 83
Figure 5.7 Publish extensions on LDAP screen .......................................................... 85
Figure 5.8 Import from LDAP screen ..................................................................... 86
Figure 5.9 LDAP server settings ........................................................................... 86
Figure 6.1 Call Detail Record .............................................................................. 88
Figure 6.2 Personal CDR .................................................................................... 89
Figure 7.1 Call Recording  Call Recording settings .................................................. 90
Figure 7.2 Local storage screen ........................................................................... 90
Figure 8.1 Troubleshooting screen ........................................................................ 91
Figure 8.2 Processing capture ............................................................................. 91
Figure 9.1 Provisioning procedure, devices configuration ............................................ 93
Figure 9.2 Adding new provisioning device ............................................................. 93
Figure 9.3 Adding new template .......................................................................... 94
Figure 9.4 Loading a configuration template ........................................................... 94
Figure 9.5 Template management screen ............................................................... 94
Figure 9.6 Built-in editor for online template editing ................................................ 95
Figure 9.7 Device/Extension/Template connection ................................................... 95
Figure 9.8 Provisioning configuration files management screen .................................... 96
Figure 10.1 PBX  File Manager  Sounds ............................................................. 99
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Tables
Table
Table
Table
Table
Table
1.1
9.1
9.2
9.3
9.4
Default values ......................................................................................9
Provisioning, Supported phone models ...................................................... 92
Placeholder available for Yealink provisioning templates ............................... 97
XML TAGs for Linksys phones provisioning .................................................. 98
Snom provisioning templates placeholders ................................................. 98
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System Management
Chapter 1 -
1.1 Introduction
KalliopePBX system can be configured through the integrated web interface, this allows the
user to modify system settings (network parameters, administration password etc.) and to
carry out those operations required to manage the PBX system.
User can access the KalliopePBX configuration interface through a web browser, using HTTP
or HTTPS protocol. In case of using HTTPS protocol the browser will show an error message
related to the security certificate, which is self-signed by an internal Certification Authority
(CA) and unknown to the browser. In order to continue, user must accept such security
certificate following the instructions.
KalliopePBX web interface is provided with two different authorization levels: administrator
level (level 10) and user level (level 3). User level can be enabled by the administrator, for
each extestion, assigning it a username and a password (set during the creation of the
extension). Username and password can be modified (both by the administrator or by the
single user) in the tab System  Gui Users. In addition, administrator can enable or disable
the access to the web interface for each single user. For further information on the PBX user
access please see paragraph 2.8.
Kalliope default IP address is 192.168.0.100. Figure 1.1 shows KalliopePBX login screen.
Administrator can login typing his password and clicking on the button ‘Administrator Login’,
while users must provide their username, password and domain associated to the extension,
and eventually click on the button ‘User Login’. Users associated to local extensions must
select “local”, in case of remote extensions’ users, it is possible to choose the corresponding
domain from a drop-down menu.
KalliopePBX login screen displays some useful information about the system: in the left corner
there are KalliopePBX Part Number and Serial Number (in this format: NR/Kxxxxxx) related to
the device is use, while on the right one it is indicated the running firmware version.

Figure 1.1 Login screen
This chapter deals with the administrator level access, user level access will be took in
consideration in another section. Here below, in Table 1.1, are the default values needed for
the first access to the web interface and the administrator default password.
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Parameter
Default value
IP Address
192.168.0.100
Netmask
255.255.255.0
Default gateway
192.168.0.1
Administrator password
admin
Table 1.1 Default values
1.2 Administration GUI
Figure 1.2 shows KalliopePBX configuration interface at administrator level, it is organized
into a double level menu.
Figure 1.2 KalliopePBX configuration interface
With the selection of a tab on the first level, it will be shown the relative submenu on the
second level.
The first level menu is composed by the following categories:
-
PBX: this menu allows the administrator to configure PBX specific features. In
particular, it is possible to create and manage: local and remote extensions, call
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groups, ACD, outbound lines (gateways or VoIP devices), incoming and outbound calls
routing, opening and closing time, custom and system audio files.
PBX Applications: this menu permits to configure advanced services such as “External
APIs” introduced with firmware release 3.9.5.
System: this menu allows a global configuration and management of the device. In
this section administrator can set network parameters, manage firmware updates and
system configuration backups, set up the High Availability system (matching the
switchboard with another KalliopePBX device) and manage additional licenses (such
as G.729 transcoding licenses).
Phonebook: this menu allows the administrator to manage contacts, available for
click-to-call service. In addition, it is possible to configure LDAP servers in order to
download external phonebooks or export local phonebooks, both to a file or to the
KalliopePBX embedded LDAP server.
Logs: this menu allows the consultation of CDR (Call Detail Record) entries stored on
the device. The administrator can consult all the PBX call records, while the user can
acccess only those records related to his own account. When Kalliope CallCenter
module is active, this menu will also show the Call Center CDR and a history of events
related to the PBX (such as CTI: login/logoff, CALL CENTER: pause/unpause,
add/remove dynamic operator, start/stop audio supervision).
Call Recording: this menu allows the administrator to set parameters related to
phone calls recording (on-demand recording, unconditional recording, recording for
incoming and/or outbound phone calls). In addition, it is possible to manage the
recorded calls stored on the internal KalliopePBX flash memory or on an external USB
drive.
Troubleshooting: a set of tools useful for troubleshooting is here available.
Provisioning: thanks to this function it is possible to create a configuration file
intended for some VoIP phones models.
File Manager: the administrator can find here some tools to upload audio files, to
convert them to G.711 and G.729 (with the purchase of a license) and to upload files
to the root folder on the embedded TFTP server.
In the top left corner it can be found the ‘Logout’ button. Next to the ‘Logout’ button there
is a news box (information concerning the release of a firmware update appears here).
The selection of the language is available in the bottom right corner, clicking on the ‘+’
button an extended list of languages is displayed. Firmware release 3.9.6 introduces Spanish
as new i7nterface language, in addition to Italian and English.
Links to the product and producer website, credits and download page are available in the
bottom left corner.
Accessing the system as an unpriviliged user, only some of the previous features are
available: CDR (only user’s own phone calls are displayed), phonebook (it includes a personal
user phonebook section) and a tab that allows to manage user’s log-in credentials.
1.3 A General Overview
In order to perform a change to KalliopePBX settings, user must follow a two-step procedure
(Save and Apply). It is possible to perform and save several changes and, eventually, apply
them with a single click on the ‘Apply’ button. Attention: not applied changes are stored in a
volatile memory, in case of an accidental reboot of the device, any change will be lost. Not
applied changes can be ignored restoring the previous configuration.
Here below an overview about incoming and outbound phone calls management.
Each VoIP phone (material ones as well as SoftPhones) must correspond to a KalliopePBX user.
Users are identified by an extension associated to the device.
VoIP devices periodically communicate their locations to KalliopePBX which keeps an updated
list. Each extension can be associated to one and only one location at a time. In case of an
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extension associated to two different devices, its location will be automatically assigned
considering the last update.
As for extensions, the configuration process is needed also for gateways (SIP-to-ISDN, SIP-toPSTN or SIP-to-GSM), VoIP devices and VoIP Trunks that allow the PBX to access the telephone
network.
KalliopePBX main task is to allow comunication between extensions, but also between an
extension and an external phone (both a traditional or a VoIP one). In addition to this,
KalliopePBX offers functions such as audioconference room, voice mailbox, call transfer and
others switchboard services.
1.3.1 Phone calls originated from a PBX extension
The number dialed by the user is managed by KalliopePBX according to an Internal Numbering
Plan which defines a set of rules in order to match the user action, as follows:
1. Telephone services
2. Speed-dial (custom and system-wide)
3. Custom selection associated to call groups, ACD, etc.
4. Local extensions
5. Remote ranges of extensions (accessible via the configurable phone trunks)
6. Outbound line reservation using a reservation code associated with a single
gateway/Trunk/VoIP Termination
7. Outbound line reservation using the LCR rules associated to the PBX user.
Internal Numbering Plan is available in the tab PBX Numbering Plan. Such screen is
divided into two columns, in the left one it is displayed the dial selections, while in the right
one the associated calls destinations. The Numbering Plan is browsed sequencially and
responds to the first match rule. The first positive match defines the call destination.
In order to reserve the outbound line, the prefix is 0 (or 0*). It is possible to omit such prefix
selecting the dedicated checkbox in the tab PBX  Outbound Routing; the switchboard will
automatically decide whether the call is an internal one or must be routed outside.
A ‘reservation line code’ can be associated to each gateway, VoIP Termination or Trunk
registered on the PBX; this numeric code must precede the called number. The phone call
will be routed using the corresponding line, in spite of the caller class and the call routing
rules (optional in firmware version 3.9.0 or greater). Thanks to this code users can overlook
default KalliopePBX routing rules.
Note: in order to allow users to place outbound calls it is necessary to
configure at least one outbound line as the default outbound call route.
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Moreover, it is possible to assign to each extension a specific LCR class consisting in a set of
LCR rules. Such rules define outbound calls routing.
If ENUM support is enabled for a specific LCR rule, the outbound line routing is preceded by
an ENUM resolution on the configured ENUM domains. If the ENUM resolution is successful, the
call is forwarded using the destination domain, or one of the configured VoIP Termination or
Trunk; on the contrary, whether this process cannot be completed successfully, KalliopePBX
will forward the call using the outbound line associated to the rule.
1.3.2 Incoming calls
The way an incoming call is managed by the PBX depends on the set of features that have
been enabled on the gateway and on the PBX itself 1, such as the DID (Direct Inward Dialing),
welcome message, opening time check, default call destination.
Incoming calls enter the switchboard through one of the available incoming lines (gateways or
VoIP Terminations 2 ). KalliopePBX verifies if a DID is defined on the incoming line for the
called number3; if a positive match is found, it is performed the action associated to the
called number. Available actions are:


Forward the call to:
o
a specific PBX extension
o
a call group
o
an ACD
o
an IVR menu or submenu
o
an opening time check
o
an audioconference room
o
a DISA service.
Forward the call to the default destination.
In case of negative match, the call is forwarded to a ‘catch-all’ destination, the available
actions for this destination are the same of the previous case.
Both for DID and ‘catch-all’ destinations, incoming calls are submitted to the rules configured
in the tab PBX  Incoming Call Management. Such rules are:

If a welcome message is configured, it is played when the call is answered, otherwise
the call is forwarded to the next step.

The call is forwarded to a destination selected from a drop-down menu.
1
For incoming calls originated from an external range of extensions please see VoIP Trunks
configuration.
2
VoIP Trunks are managed differently, the called number is forwarded to the Internal
Numbering Plan; exceptions to such behavior can be configured in the DID section.
3
A single gateway can, indeed, be associated to several geographical numbers; for instance
when an ISDN PRI line or multiple ISDB BRI access are available.
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One of the possible destinations is the opening check time; it is organized into a two-steps
process:

The actual time and date of the incoming call is compared with a set of days and time
defined as closing days; it gives the user the possibility to set holidays and special
events not weekly recurring.

The actual time and date of the incoming call is compared with a set of days and time
defined as weekly closing days.
When a closing date and time is matched, the user has the possibility to configure an audio
message (a backdoor code to force the call can be configured) and eventually performs one of
the following actions:
o
Hang up
o
Forward the call to an extension
o
Forward the call to a call group (with or without a different opening time
check)
o
Forward the call to an ACD (with or without a different opening time check)
o
Forward the call to a user voice mailbox
o
Forward the call to an IVR menu
o
Forward the call to an external phone number (availability), it is also possible
to configure the gateway used for the outbound calls
o
Forward the call to another opening time check.
When no closing time and date is matched, the incoming call undergoes a specific action
configured from a dropdown menu similar to the previous one.
Note: call routing can be redefined for each number available on a gateway
(or VoIP Termination) through the setting of the DID. In this case, the call is
forwarded to the configured destination (call group, ACD, extension, IVR)
passing over the incoming calls rules and the welcome message.
Callers’ numbers can be preceded (optionally) by the prefix 0*, in order to give the user the
possibility to recall them directly from the call history on his phone.
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Chapter 2 -
System Menu
2.1 Status
Figure 2.1 System  Status
The tab ‘Status’ offers an overview about KalliopePBX, in particular: Model and Serial Number,
non-volatile storage (free space on the disk), current firmware version and available updates,
CPU load and temperature4, and current time.
2.1.1
Firmware version
For a detailed description of KalliopePBX firmware management please see ‘Firmware
Update’.
2.1.2
Date and time
Time is synchronized using NTP, a set of geographical servers or user’s custom time (editable
in System  Network). It is possible to force synchronization by clicking on the button
‘Synchronize now’. It is also possible, by clicking on the pencil icon, to set a temporary
different time and date, useful to test opening check time, for example. Time will be
automatically reset through NTP after five minutes, as shown in Figure 2.2.
4
Such information is available only with KalliopePBX-V3.
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Rev.: 15-02-2015
Figure 2.2 Manual date and time setting
If the NTP server cannot be reached, user time and date settings will remain operational until
any further modification.
From firmware release 3.9.6 it is possible to set the time zone from a drop-down menu. The
change is immediately operative, the page must be refreshed in order to display the updated
time.
Note: it is necessary to check periodically system date and time since
closing time check depends on their correct setting. In case of a
switchboard without any access to NTP, the system clock can show a
considerable gap from the correct time. It is suggested to check weekly its
accuracy.
Concerning the disk space, KalliopePBX uses it to store CDRs, audio files (updated or recorded
by the user), configuration backups and voice mailbox messages.
2.1.3
VoIP Services
In this frame there are two buttons: the first allows the user to force the reload of PBX VoIP
configuration (without interrupting the ongoing services), while the second allows the user to
force a reboot of the telephone network (the ongoing calls will be aborted).
From firmware release 3.12 it is possible to reboot the telephone engine using the “escaped
mode”: the character # (if present) of the selection sent by KalliopePBX to a gateway or a
VoIP provider is replaced by the escape sequence (%23). This change requires the reboot of
VoIP service, so that the ongoing calls will be interrupted.
2.1.4
SIP PnP Service
In this frame it is shown the daemon SIP PnP execution status. This feature is useful to
automatically manage devices provisioning without any configuration on the DHCP server;
when the daemon is running, PBX listens on multicast address 224.0.1.75:5060. When a
telephone boots, the SIP PnP process requires it to send a message SIP SUBSCRIBE to this
address. The message needs to include its trademark, model and MAC address. PnP service
running on KalliopePBX receives such message and (for supported models 5) replies with a SIP
NOTIFY message providing the phone with the IP address needed to download its
configuration file created by KalliopePBX provisioning system.
SIP PnP service can be manually launched or stopped by clicking on the button next to the
status indicator. Changes are immediately applied, but not persistent after a reboot
(automatic launch can be configured in the tab System  Network Services).
5
Current supported models: Snom (MAC Addr: 00:04:13:xx:xx:xx), Gigaset Pro (MAC Addr:
7C:2F:80:xx:xx:xx) and Yealink (MAC Addr: 00:15:65:xx:xx:xx).
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2.2 Network
Figure 2.3 System  Network
In the tab ‘Network’ user can configure settings related to IP addresses, DNS and routing.
With firmware release 3.9.3, multiple network interfaces are supported (available only for
models with multiple physical network interfaces such as KPBX-V3-R, KPBX-V3-R-OPT-FO,
KPBX-V3-MED800 hosted within the OSN modules of Audiocodes Mediant 800 devices and for
virtual appliance KPBX-V3-ESX which supports an arbitrary number of interfaces).
In the frame ‘Network Configuration’ are shown available interfaces and, for each of them, it
can be assigned a satic IP address (and relative subnet mask) or it can be enabled the
automatic IP assignment via DHCP.
Moreover, it is possible to configure DNS server’s IP address, as well as one or more static
routes. Further than the default route (0.0.0.0/0), an arbitrary number of routes can be
configured (in the compact format x.x.x.x/NN) specifying, for each of them, the related
gateway. PBX system checks whether the gateway belongs or not to one of the connected
subnets, if not it prevents the input of such entry.
Whether High Availability option is enabled, ‘Network Configuration’ frame is available in
read-only mode; in order to modify such values it is needed to disable High Availability option
since any change could break the coherent configuration of HA nodes.
Note: an erroneous configuration of network parameters (IP address and
subnet mask) may cause the unavailability of the PBX interface. Please
double-check the accuracy of these values before applying any change.
Note: it is available a software tool (Kdiscovery) useful to scan LAN and find
any KalliopePBX installed, as well as their IP addresses, firmware versions,
etc.
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2.3 Network Services
In this tab several parameters related to network services can be configured.
Figure 2.4 System  Network Services
In the first frame it is possible to manage PBX clock synchronization choosing between default
NTP servers (NTP server randomly chosen from [0,1,2,3].debian.pool.ntp.org) or a specific
server, providing IP address and the corresponding radio.
The following checkbox enables or disables automatic launch of SIP PnP Service at PBX boot.
In the SMPT frame it is possible to configure SMPT parameters needed by KalliopePBX to send
e-mail messages (in particular voice mailbox notifications, with or without attachment, and
administrator notifications, for example an available firmware update).
Administrator e-mail address is used by KalliopePBX both, as addressee for notification
messages and as sender, except for those related to voice mailbox whose sender is specified
in PBX  Features. Other parameters required are:

SMTP server name or IP address and port (default port is 25)

SMTP credentials (username and password) to log on SMTP server (only AUTH LOGIN is
supported)

Optional support for STARTTLS (when available on the server) to encrypt delivering
mail.
The button ‘Test e-mail configuration’ allows to test the correct functioning of configured
settings. E-mail testing uses the current operative configuration on KalliopePBX, in case of
unsaved changes or saved but not applied changes, the system will not take them into
consideration.
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2.3.1 NAT Traversal via external IP
NAT Traversal can be useful, for example, when the switchboard is on a private local
network, behind a NAT, and uses a VoIP Trunk of an operator whose VoIP platform does not
correctly support NAT Traversal. Another example could be when KalliopePBX needs to access
phone extensions that are out of the NAT (configured with a public IP address, while
KalliopePBX is configured with a private IP address).
To enable this feature, it must be provided the external public IP address for which the NAT
is realized. The activation of the NAT Traversal entails all SIP signals from KalliopePBX to be
modified as if the switchboard were configured with a public IP address (IP addresses of
different SIP URI, IP addresses of different RTP media flows specificied in SDP, etc).
Figure 2.5 System  NAT Traversal
Saving the NAT Traversal configuration will automatically configure the localnets, that are
exceptions to this rule for which KalliopePBX will not modify the SIP signals (since they
communicate directly with the PBX and do not use the NAT system).
It is also possible to configure further localnets in order to manage additional subnets
(accessible by KalliopePBX through routing) for which the configuration of a public IP would
prevent any communication with PBX.
2.4 SNMP
This tab allows the user to enable or disable the integrated SNMP agent and to modify the
community string used to read data. In addition, it is possible to modify the ‘System Location’
and ‘System Contact’ OIDs. KalliopePBX integrates the standard Net-SNMP agent.
Figure 2.6 System  SNMP
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2.5 Firmware Update
Figure 2.7 System  Update
Through this menu it is possible to verify the availability of a new software release, to
download from the server and to install it. The screen shows the firmware version in use and
a list of updates available for download.
In case the most recent release is unknown, it is possible to click on the button ‘Check for
updates now’ to download a list of available updates. In addition, KalliopePBX can
automatically check for updates (daily or weekly) when this option is enabled.
Note: by clicking on ‘Save settings’ the automatic updates check is
immediately enabled.
Normally, firmware updates must be installed sequencially, that is, the installation of a
recent update is not allowed whether the previous versions are missing.
When several updates are available, it is suggested to install the most recent one. Whether
an older version is needed to update to a newer one, the latter will appear in grey and a
notification will be displayed. Usually, within the same major release, it is not necessary to
install each intermediate version (for example, if 3.6.0 is running on the PBX it is possible to
update to 3.6.6 without installing 3.6.1). On the contrary, to update to a different major
release, a necessary step is required (for example, if 3.5.5 is running on the PBX it is not
possible to update to 3.6.6 without installing previously 3.6.0).
Whether ‘High Availability’ option is enabled, it is possible to check for updates but not to
install them in order to prevent a possible incoherence between the HA cluster nodes. To
complete a firmware update, ‘High Availability’ option must be previoulsy disabled.
2.5.1
Update Process
KalliopePBX is provided with two non-volatile storage areas dedicated to firmware files. Such
areas contain the main firmware and the secondary one.
Normally, the main firmware is the one in use. When a new release is available, the new
firmware is installed on the secondary area. It is suggested to backup and download the
configuration files before proceeding with the update. KalliopePBX automatically stores a
configuration backup in a dedicated area.
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When the update process is complete, it will be displayed a button to reboot the PBX with the
secondary new firmware (please see Figure 2.8). Whether the boot with the new firmware
causes problems, a simple reset of the switchboard will make the PBX boot with the old
firmware version still stored in the main area.
Figure 2.8 'Reboot with secondary firmware' button
Normally, the reboot with secondary firmware is successful. Logging in as administrator,
KalliopePBX will display a message notifying that secondary firmware is running; it is
suggested to check the correct functioning of the system before completing the update
process.
It is possible to decide whether confirm the new update as the main one or restore the
previous version. When the user decides to apply permanently the new firmware, a reboot is
performed; the secondary firmware is now the main one, while the previous version is stored
as secondary firmware.
Note: if the High-Availability option is enabled the update is not allowed.
Please disable the High-Availability option and proceed with the update;
after a successfull update process the High-Availability option can be reenabled.
Please note: it is possible to restore the previous firmware version in any moment by clicking
on the button ‘Reboot system using secondary firmware’ (System  Status) and selecting this
choice as permanent. Main firmware and secondary firmware are associated to their own
configuration files; if secondary firmware is restored, its last configuration is recovered.
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2.6 AMI
It is possible to integrate external applications that use AMI (Asterisk Manager Interface). This
panel permits to define the authentication credentials and an ACL constituted by one or more
IP addresses/subnets.
Figure 2.9 System  AMI
The defined user has read permissions “call” and write permissions “call,originate”.
The AMI port is 55038 and not the standard 5038.
2.7 Configuration Backup
Figure 2.10 System  Backup
Configuration backups can be created through the relative tab; these are stored on
KalliopePBX memory disk but it is suggested a download. Backups are associated to the
firmware version in use; it is not possible to restore a backup created on a firmware version
different from the one in use.
Note: remember to create a configuration backup each time a change
and/or a firmware update occur (backups created with previous versions
are not compatible with a new one).
Backups can include phonebook and/or custom audio files uploaded on KalliopePBX.
Moreover, a short annotation can be added, it will be shown when restoring the backup. If
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custom audio files exceed the available space on disk, backup process could fail and no
archive will be created; if so, custom audio files can be manually downloaded from the tab
‘File Manager’. Once restored the configuration backup, such files can be uploaded and, if
mantaining the same name, they are automatically recognized by the PBX.
Restore process can be launched by clicking on the button ‘Restore Backup’ associated to the
desired backup.
Figure 2.11 Restoring configuration backup
A window containing some backup details is shown (date and time, user annotation, firmware
version, saved data); user can specify what to restore or not. Clicking on ‘Restore Backup’ the
configuration file is loaded and shown to the user, but not applied. Administrator can check
and, possibly, modify saved settings before making the configuration backup operative.
During this phase, the access at user level is disabled (an explicative message is shown), while
an alert warns the administrator that a restore operation has not been completed and should
be finalized or aborted (Figure 2.12Errore. L'origine riferimento non è stata trovata.).
Figure 2.12 Alert of uncomplete restore process
As a final step of this process, the configuration backup is saved and made operative.
NOTE: when configuration backup is loaded but not restored, phonebook and custom audio
files shown are not those included in the backup file, but those in use at the moment of the
launch of the restore process. Backup phonebook and custom audio files will be shown when
the restore process is finalized.
2.8 KCTI/Web GUI Users
In this tab administrator can manage users’ credentials (needed to login the GUI and
KalliopeCTI application). Administrator can enable or disable access for specific users, assign
them Kalliope applications licenses (such as KalliopeCTI Pro or Phone and the new Kalliope
Operator Seat), assign Call-center role (operator or supervisor) if the corresponding license
has been activated on the PBX.
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When a new extenstion is registered (both local or remote), it is automatically added to the
list shown in this panel. For local extensions, default credentials are the same as SIP
credentials (username and password) defined in the extensions tab. For remote extensions no
password is assigned and the web access in disabled; in order to enable web access for
remote extensions, a password must be assigned by the administrator.
Figure 2.13 System Web GUI Users
Figure 2.14 System  User access credentials screen
For each extension (both local or remote), either administrator than user can modify
credentials. When changing a username, CDR and personal Phonebook keep linked to the user
extension.
Figure 2.13 shows a list of extensions; for each extension it is displayed status, username and
assigned licenses. For remote extensions it is also shown the trunk domains used to reach
them; each remote user must select its trunk domain from a drop-down menu when logging
in.
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To modify users’ credentials (Figure 2.14), administrator must click on the modify button. In
this screen administrator can: modify username, modify or assign password, assign editing
privileges concernign the Shared Phonebook and assign the licenses to enable the use of the
relative applications. Changes perfomed in this screen are immediately operative, no save or
apply is needed.
When KalliopePhone is running, PBX automatically performs phone SIP provisioning with the
credentials of the extension linked to the KalliopeCTI user.
A license can be assigned only to one user at time (KalliopePhone licenses are not linked to
the user, but do not admit a simultaneous use).
2.9 Administrator Settings
Figure 2.15 System  Settings
PBX administrator can change WEB GUI admin access password (username cannot be
modified). In order to complete such operation, old password and new one must be provided,
clicking on ‘Save’ button the process is finalized. In the same way privacy password can be
modified, such password is needed to disable masking for telephone numbers stored in the
CDR tab.
Note: clicking on ‘Save’ button changes are immediately operative.
2.10 High Availability
High Availability (HA) option permits to connect two KalliopePBX switchboards making them
operate in 1+1 redundancy, in Active-Backup mode. Changes performed on the active PBX are
syncronized on the backup PBX, including CDRs. In case of a fail of the active PBX, the backup
PBX automatically becomes the active one; settings and data are restored in few seconds.
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During the switch, phone calls are aborted while devices registration keeps functioning in
order to guarantee a fast recovery of the routing activity.
HA configuration requires a pairing process between two KalliopePBX devices. Pairing process
can be achieved with a KalliopePBX FO kit (FailOver, a master full-function node plus a slave
node) or with two KalliopePBX master devices.
Slave devices can operate only as a part of a HA cluster. Such devices have a limited WEB
GUI: only administration and HA paring functions are enabled. In case of a fail in the cluster,
slave device keeps working regularly for a month, after such period slave PBX stops working
until a new pairing process is performed.
When HA pairing is performed between two master PBX devices, in case of a fail of the active
PBX, the backup one will keep fully working without any time restriction.
2.10.1 Enabling HA
To enable HA function, PBX administrator must click on the ‘Activate’ button in the tab
System  High Availability. Once the HA option is enabled, the cluster can be configured on
one of the nodes. Configuration is automatically transferred to the second device during the
pairing process. In order to perform a successful pairing process, the two PBX devices need to
run the same firmware version.
Figure 2.16 HA configuration screen
The configuration sender node is called ‘North’, the receiver one is called ‘South’. During the
pairing process, configuration settings (including CDRs and Phonebook) on the ‘North’ node
are transferred to the ‘South’ node overwriting any previous configuration.
In the following example, HA is enabled only on the ‘North’ node. HA configuration requires:
‘South’ node serial number and IP address, and cluster IP address 6.
Both nodes need to be on the same Layer 2, HA cluster IP address must pertain to the same IP
subnet of the ‘North’ node. In case of two physical PBX devices equipped with a serial port, it
is suggested to use a null-modem serial cable (DB9 F/F serial cable included in –OPT-FO
models).
HA configuration progresses through different steps:
6
This is main difference with 3.9.2 firmware releases and older. In such firmware releases HA
IP address was set to the coordinator node address and a new IP address had to be assigned to
the coordinator. Starting from firmware release 3.9.3, resources (SIP engine, web GUI, KCTI
service) are available on an IP different from the nodes ones, so that each node is always
reachable at its own IP address whatever tha HA status is.
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Rev.: 15-02-2015

‘Enabled’ checkbox: indicates if HA is enabled on that specific device

‘Sync’ radio button: indicates the devices involved into data exchange and
syncronization

‘Default active’ selection: it gives the user the possibility to choose on which node HA
option must be default active7

a list of IP addresses must be provided, each node uses them to test network correct
functioning.
Clicking on ‘Save and start HA’, the HA service is enabled on the ‘North’ node; the screen will
show the cluster status. Until the pairing is not complete the ‘North’ node WEB GUI can be
reached through its IP address, once the HA cluster is operative such node can be reached
through the HA cluster IP address.
In Figure 2.17 HA is not enabled on the ‘South’ node, indeed the corresponding indicator is
red. ‘North’ node has not acquired the ‘South’ node yet, in fact both nodes are shown as
‘inactive’.
Figure 2.17 HA status screen, acquisition in process
The acquisition process takes few seconds, once it is concluded, the ‘North’ node appears as
the active node (please see Figure 2.18). The black marker above one of the nodes indicates
which WEB GUI is in use.8
Figure 2.18 HA status screen, acquisition complete
7
8
In previous releases, HA was always active on the node with lower Serial Number.
Useful in case of access through the node IP address.
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Since HA option is not enabled on the ‘South’ node, pairing process cannot be successfully
finalized as shown below in Figure 2.19.
Figure 2.19 HA status screen, pairing is not possible
It is now possible to enable the HA service on the ‘South’ node; configuration is automatically
imported from the ‘North’ one.
Figure 2.20 HA status screen, HA is enabled on South node
‘Execute Pairing’ button appears now, clicking on it pairing process is launched: HA
configuration and PBX configuration are transferred from ‘North’ to ‘South’ node and nonstop syncronization between the devices is started.
As shown in Figure 2.21 pairing in now in progress.
Figure 2.21 HA status screen, ongoing pairing
When pairing process is complete, ‘South’ node HA status indicator turns green and the
cluster becomes operative. ‘Eth Link Status’ indicates that data exchange between nodes is
functioning in a correct way.
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Rev.: 15-02-2015
Figure 2.22 HA North node status screen, pairing process complete
‘Go Standby’ button forces a switch between ‘North’ and ‘South’ nodes, the latter becomes
now the active node. Such function is useful in case of the active node needs device
maintenance.
The ‘Acquire resources’ button, in Figure 2.23, permits the ‘South’ node to acquire resources
from the ‘North’ one, this operation turns the former into the active node and leaves the
latter on standby.
Figure 2.23 HA South node screen, pairing complete
2.10.2 Disabling HA
It is possible to disable HA function on both, active or standby node.
Disabling HA on the standby node, service is disabled only on such node; the active node
keeps on working normally, no disruption will affect the system. The HA cluster is demoted
but still running, such operation can be useful in case of device maintenance or replacement;
when standby node is reactivated, HA service must be manually enabled. Disabling HA on a
backup node, VoIP services on such node are aborted, an alert message is displayed and a
device reboot is needed.
Note: reboot does not affect HA enabled status, when nodes boot again
last HA configuration is recovered. If HA is enabled on both nodes and they
are not able to communicate each other, both nodes could acquire
resources simultaneously.
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Disabling HA on the active node, HA service is aborted on both nodes. Nodes are now
independent and the cluster is disabled; to reactivate the cluster it must be followed the
enabling procedure (please see paragraph 2.10.1).
2.11 Licenses
Figure 2.24 System  Licenses
In this tab G.729 license (available on Digium Store) and KCTI Pro/KalliopePhone licenses can
be activated.
2.11.1 G.729 Licenses
For each G.729 license several information is displayed:

License Activation Key

Licensed channels

License expiration date

License Activation Status
Licenses files are available for download; in case of a device fail, such files can be uploaded
with no need to reactivate the licenses.
Clicking on the button ‘Activate New License’, license activation page is displayed, two
sections must be filled: ‘Digium License’ and ‘Personal Info’.
In the ‘Digium License’ section it must be provided:

Product Type: user must select, from a drop-down menu, which product license
intends to activate.

License Activation Key: alphanumeric key previously provided by Digium.
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Figure 2.25 G.729 License Key Activation screen (Digium License)
In the ‘Personal Info’ section user information must be provided, as shown in Figure 2.26.
Figure 2.26 G.729 License Key Activation screen (Personal Info)
In order to successfully complete license activation procedure, please check Digium server
availability clicking on the dedicated button9; the license can be now activated through the
button ‘Activate Now’.
Note: when HA is enabled, each node must be equipped with its own G.729
license. Licenses are associated to the specific device and cannot be
transferred.
9
Server could be displayed as unreachable even when it is actually available. License
activation could be correctly completed even when ‘Server is unreachable’ message is
displayed.
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2.11.2 Kalliope Licenses
KCTI Pro and KalliopePhone licenses are distributed as ‘.lic’ files; such files are associated to
KalliopePBX Serial Number. When a valid license file is uploaded, it automatically appears in
the available list of licenses.
A free KalliopePhone license is pre-activated on each KalliopePBX, administrator can assign
such license to one user at a time.
From firmware release 3.10, it is available a new licensed module called Kalliope Call-Center:
such module adds new functionalities specifically developed for call-center needs:

Advanced management of operators’ status for each specific queue

Dynamic operators (managed by Supervisor users)

Automatic callback on busy operators, this allows the caller to reserve a callback by
typing the key 5 and providing his phone number

Support for the application Kalliope Supervisor Panel (available on Windows) that
allows to access all the statistics related to the queues and operators and perform
actions like adding or removing dynamic operators from a queue

CallCenter CDR, that is, an advanced log that contains details on incoming calls
managed by waiting queues such as Total Waiting Time, Conversation Time, Operator
who answered the call etc.

With firmware release 3.11 it has been added the fuction Spy/Whisper/Barge
supervision: by configuring a service’s prefix in the Numbering Plan, the Supervisor
users can supervise operators dialing the prefix plus the operator number. The
supervision’s begin/end is notified to the operators by a customizable audio message.
Supervisor users can choose between three types of supervision: Spy (key 4) Whisper
(key 5) and Barge (key 6). The supervision activity events are saved in the
KalliopePBX logs and are available from the panel “Logs”  “PBX Events”.
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Chapter 3 -
PBX Menu
3.1 Extensions
Figure 3.1 PBX  Extensions
‘Extensions’ tab allows the administrator to define local extensions (username and secret SIP
identify each extension) and associate them an identity and relative authorizations. It is also
possible to manage remote extensions, that is, those external ranges of extensions accessible
via VoIP trunks.
‘Extensions’ tab is subdivided into three sections: local extensions global settings are shown
in the first one, while lists of local and remote extensions are displayed in the other two.
3.1.1 Global Settings
Figure 3.2 PBX  Extensions  Global Settings
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In the ‘Global settings’ section several parameters, valid for every extension, can be defined.
Such parameters are:





Default ringing timeout: defined in seconds (valid for calls coming from extensions or
from outside the network). This value is not valid for calls addressed to call groups
and ACD (which have their own ringing timeout, please see 3.1.3.1).
Expiration maximum time allowed for extensions registration: defined in seconds and
valid for every extension. With small values, extensions send a registration request
more often; this permits to reduce unavailability time after a PBX reboot. This value
also affects ACL; in fact, ACL is applied only after a registration request (if an
extension IP address is not included in an ACL, such extension can continue to make
phone calls until its next registration).
Enable or disable return call to transferer if transfer destination is busy, not available
or not answering (only if failover action configured for that extension is ‘hang up’,
otherwise the transferred phone call will behave as configured as failover action).
ACL (Access Control List): this function permits to define a subnet (IP address and
netmask) from which KalliopePBX accepts extensions’ registrations. Such ACL can be
modified for each extension.
Electronic lock: with the firmware release 3.9.6 it has been added the possibility to
assign an additional LCR class, valid only when the extension is locked. This new
feature permits to place calls to pre-defined numbers (such as emergency or
availability ones) even if the extension status is locked. By checking the relative
checkbox such function is globally enabled; two parameters can be set: Unlock mode
and Unlock policy. This configuration is valid for every extension; nevertheless, each
extension configuration can be manually modified.
Unlock mode indicates if extensions: are able to place a call (Open), need to insert an
unlock code (default value is 850) or need to provide a password (Password). Unlock
request permits to prevent unauthorized clients (such as dialers) to place calls.
Unlock policy indicates for how long the unlock status is valid. Possible values are:
‘One shot’, unlock code or password must be provided each time a phone call is
placed, ‘Relock after xx minutes’, a custom timeout value can be defined, ‘Unlocked
until the peer lock it again’, the extension is unlocked until the lock code is typed
(default value is 851).
Any change applied to these parameters will cause a reset of extensions lock
configuration.
Figure 3.3 Extensions unlock screen

Failover actions: permits to define the failover action to perform in case of:
o
o
o
No answer, ringing timeout exceeded;
Busy;
Unavailable, usually when the device is not registered on the PBX.
Phone calls are, in addition, divided into incoming and outbound calls so that, on the whole,
there are six different failover actions available.
A message can be optionally played to the caller, possible failover actions are:
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







Hang up;
Forward to
Forward to
Forward to
Forward to
Forward to
Forward to
Forward to
a specific extension;
a call group (with or without the relative opening time check);
an ACD (with or without the relative opening time check);
a voice mailbox;
an IVR menu or submenu;
an external number, specifying the routing gateway to use; 10
an opening time check.
When the failover action is ‘Hang up’, ‘Return to transferer’ option may be enabled for phone
calls directly transferred to such extension. In this case, the call will return to the transferer
and the number on the display will be preceded by an ‘R’.
Figure 3.4 Failover actions configuration screen
3.1.2 Local Extensions
Local extensions section shows the number of defined extensions, the number of licensed
ones and the maximum number of extensions admitted (in case of a limited version of
KalliopePBX).
In the same form is available the following information for each extension:






10
Status (enabled or disabled)
Extension Number
Name and Last Name
Extension LCR Class
Call-pickup group associated to the extension
Account registration status (red=not registered, green=registered)
In this case caller ID displayed will be the extension number.
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


PBX and VoIP device Round-Trip Time (RTT), useful in case of troubleshooting activity
Device IP address and port
User Agent, normally includes device model and current firmware version, it may also
include device MAC address.
Checkboxes on the left side of the screen allow the user to perform a multipe selection in
order to apply a specific action (available from the drop-down menu at the bottom of the
page) to more extensions at once. In addition, two shortcuts permit to add a new extension
or to import a list of extensions from a CSV or XLS file.
3.1.3 Add or modify a local extension
To add a new extension just click on the ‘Add new extension’ button at the bottom of the list
of extensions. To modify an already existing extension just click on the edit button (pencil
icon) or directly on the extension number. In both cases it will be displayed an extension
configuration screen; such screen is divided into several frames.
When adding a news extension, it is possible to clone the configuration from an existing one;
this procedure is useful to speed up the creation of multiple extensions that share the same
basic configuration.
3.1.3.1
Account
Figure 3.5 Extension configuration screen, Account
Account frame allows the user to modify several parameters:


Extension Number: this entry only admits numerical values, it cannot be modified
once user creation process is finalized. Extension number corresponds to the login
username (also called UserID or AuthenticationID) needed to register the device on
the PBX.
Password: this entry admits alphanumerical values, between 3 and 30 characters.
Password is needed to login when registering the device on the PBX.
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Note: it is important to adopt an extremely complex password in order to
prevent brute-force attacks by automated dialers (external or internal to
the local network) and unauthorized calls.






Maximum concurrent calls: this value refers to the total amount of simultaneous call
admitted for such extension (both incoming or outbound). If this value is 1, call
transfer is not possible; in order to allow call transfer the minimum value to set is 2.
To return a busy tone when the extension is already engaged in a conversation, the
Call Waiting option must be disabled.
Busy level: it indicates the number of calls admitted before the extension returns a
busy tone. It is useful, for example, to disable the Call Waiting service by setting this
value to 1. In case of secondary devices configured, this field indicates the maximum
number of calls admitted by all the devices.
ACL: allows to overwrite global ACL configuration for such extension.
Extension Lock: allows to overwrite global lock configuration for such extension
(global lock/unlock extension function must be enabled).
Disable NAT support for this peer: this checkbox allows to disable NAT traversal for
such extension. It is suggested to keep this function unchecked, but, for some
telephone models, it must be selected in order to register the device on the PBX.
Disable: this checkbox permits to temporary disable such extension; specific
extension configuration is preserved.
3.1.3.2 Secondary devices
Multiple SIP devices can be associated to the same main identity. Such devices have their own
SIP credentials and account, but they share the identity of the main extension. Outbound
calls placed by secondary devices will show the main identity number and, consequently,
incoming calls are forwarded to all the secondary devices with a ring-all policy.
In addition, it is possible to set a “busy level” to the main extension so that, when this level
is reached, the extension returns a busy tone. The main extension BLF status considers the
global use of all the associated devices.
Each extension can have multiple secondary devices; for each device the system creates a
KCTI/Web user so that KalliopeCTI licenses can be assigned (in fact, the secondary device
could be a KalliopePhone).
The list of secondary devices configured is available from the specific tab. In terms of
configuration, secondary devices inherit all the settings from the main extension, except for
ACL and codecs.
3.1.3.3
Identity
Figure 3.6 Extension configuration screen, Identity
Identity frame contains information about the user associated to such extension, in
particular: Name, Last Name, Organization, Unit and Mobile Number. This information will be
available on the extensions’ shared Phonebook. Mobile Number is user’s external number
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(usually a mobile number); FastXfer service uses such number to transfer phone calls from the
extension to the mobile device and vice versa using a custom DTMF code.
The options ‘Edit current user’ and ‘Create a new user instead of the current one’ affect
Phonebook and CDRs; if a new user is created, previous Phonebook and CDRs will be deleted.
3.1.3.4
Publication on Phonebook
Figure 3.7 Extension configuration screen, Publication
Available publication options are:




Disabled
Enabled, with default LDAP settings rule
Enabled, publishing the number in the box below
Enabled, with default LDAP settings rule applied to the number in the box below
(masking)
3.1.3.5
Call-pickup group
Figure 3.8 Extension configuration screen, Call-Pickup Group
This frame allows the creation of a group of extensions with the authorization of call-pickup,
typing the pickup code *9 (followed by Enter button). The call-pickup group operates as
follow: each user/extension belongs to a call-pickup group (by default all users belong to
group 1) and is authorized to pickup incoming phone calls addressed to a member of such
group; defaul call-pickup code is *9 but it can be modified in PBX  Numbering Plan. Each
extension can pickup phone calls addressed only to his group.
Note: *9 code must be followed by the press of the Enter key on the phone
(or wait for the digit-timeout set on the phone).
Note: if multiple calls are ringing on extensions belonging to the same
pickup group, *9 will pickup the most recent call.
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3.1.3.6
Outbound calls
Figure 3.9 Extension configuration screen, Outbound calls
‘Outbound calls’ frame allows the user to assing two LCR classes (a primary one and a
restricted one) to each extension; a set of LCR rules manage outbound calls routing. LCR class
can be configured in PBX  Outbound Routing.
The restricted LCR class is valid when the extension status is locked and it is useful to place
calls to pre-defined numbers (such as emergency ones).
When the electronic lock is disabled, or the extension status is unlocked, the use of a
gateway prefix bypasses the LCR routing configuration; users can, then, use a specific
gateway overlooking the extension LCR rules. Whether the extension status is locked, the use
of a gateway prefix permits to place a call only if the restricted LCR class admits it.
3.1.3.7
Voice mailbox
Figure 3.10 Extension configuration screen, Voice mailbox
In the Voice mailbox frame following parameters can be configured:



Enable: this checkbox allows the user to enable voice mailbox for the spefic
extension.
Password: this entry only admits numerical characters. Users need this code to access
their personal voice mailbox; the access can be allowed from inside or from outside
the network.
Notify new voice mailbox messages by e-mail: if enabled users’ email address must be
provided. Discretionary, the voice message can be attached (as .wav file); in this case
it is possible to configure the PBX to delete the voice message after sending the
email. If this option is not enabled, voice messages are stored into users’ mailbox;
users can access their mailbox anytime, from a local device or from an external one
(if correctely set in PBX  Features).
Note: voice mailbox status indicates weather or not the user has an
associated mailbox; an enabled status does not imply that missed calls to
such extension will be forwarded to its voice mailbox. The actual
forwarding to voice mailbox must be configured as failover action in case
of missed calls for that specific extension.
In order to access their own voice mailbox, users need to dial 801 (access code to voice
mailbox service, it can be modified in PBX  Numbering Plan). Users must provide mailbox
number (the same as the extension number) and password in order to listen, delete or store
voice messages.
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Note: by dialing 801, followed by the voice mailbox number (for example
801 100), user will be required to enter the password of the related
mailbox.
3.1.3.8
Failover actions
Figure 3.11 Extension configuration screen, Failover actions
In this frame it is possible to re-define the failover actions to perform in case of busy, not
answered call or not available user, overlooking what has been configured in the extensions’
‘Global Settings’ panel.
3.1.3.9
Busy Lamp Field
Figure 3.12 Extension configuration screen, Busy Lamp Field
Through this frame it is possible to manage Busy Lamp Field function, that is, line status
indicator; enabled devices can have a visual overview of busy extensions. Extensions line
status can be displayed on one or more groups, selecting the relative checkbox. In order to
successfully use this function, telephone devices must be correctly configured.
Usually, a green LED associated to the relative extension indicates its free status, a red
flashing LED indicates that the device associated to such extension in ringing, a permanted
red LED indicates that the line is busy in another conversation.
With compatible devices, PBX allows the pickup of a ringing line by typing the code
**<extension>11. Such call pickup service does not take into consideration ‘call-pickup groups’
policy.
11
** code is set by default, but it can be modified in PBX  Internal Numbering Plan.
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3.1.3.10 Codec Selection
Figure 3.13 Codec Selection
The selection of the ‘Enable custom codec selection’ checkbox allows the user to overwrite
global codec policy (PBX  Features) for that specific extension. In case of incoming calls,
PBX will follow the codec preference order configured in this frame. Whether the device
supports video calls, user can also configure video codec preference order.
3.1.3.11 Call Recording
Figure 3.14 Extension configuration screen, Call Recording
In this frame call recording settings can be configured; such function can be enabled both, for
incoming and for outbound calls, on request (by typing *1 to start and stop recording) or
uncoditionally (PBX records the entire phone call).
Note: call recording function must be previously enabled in global settings.
If, for a specific extension, a globally not enabled option is selected, an
alert message will be displayed and such feature will not be operative.
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3.1.4 Importing custom lists of extensions from CSV/XLS files
Figure 3.15 Extensions import process
Lists of extensions can be imported from a CSV/XLS file by clicking on the specific button. A
CSV/XLS file, including the following columns (not necessarily in the same orden), is needed
to complete such process:
• EXTEN: extension number (required)
• SECRET: authentication password (required)
• GIVENNAME: user name (required)
• SURNAME: user last name (required)
• O: Organization or Company
• OU: Office or Unit
• VM_ENABLE: voice mailbox (enabled or not)
• VM_PASSWORD: voice mailbox password
• VM_NOTIFY: e-mail notification for new voice messages (enabled or not)
• VM_EMAIL: e-mail address for new voice messages notifications
• VM_ATTACH: attach audio messages to e-mail notifications (enabled or not)
• VM_DELETE: delete voice messages if attached (enabled or not).
For those fields that only admit boolean values, user must type 1 if enabled and 0 (or empty)
if disabled. In addition, in case of CSV files, user can specify which character used as
separator.
Clicking on ‘Choose File’ button a file browser is displayed; user must select the extensions
file to upload, specify if it is a CSV or a XLS, and, finally, click on ‘Upload’. In order to check
for errors, a recapitulation of the uploading extensions is shown.
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Figure 3.16 Extensions recapitulation screen
Whether one or more extensions are already registered on KalliopePBX, user can decide how
to proceed: already existing extensions can be updated or overwritten. To finalize the
importation just click on the ‘Import Now’ button.
3.1.5 Remote extensions
Through this form user can assign extension identity and login credentials to some (or all)
remote telephone numbers belonging to a Trunk.
Figure 3.17 List of remote extensions
For each telephone number it must be defined: extension number, identity (Name Last Name,
Office or Unit, e-mail address) and specific Trunk.
Figure 3.18 Remote extensions editing screen
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Remote extensions’ information can be shown on local Phonebook, by selecting the specific
checkbox, or on LDAP Phonebook, specifying publication policy. Moreover, a specific LCR
class can be assigned to each remote extension, overlooking general trunk LCR settings.
3.2 Call Groups and Automatic Call Distribuition
Incoming calls (both from local extensions or from external lines) can be addressed to two
different types of extensions groups, each with specific features and configuration:

Call Groups

ACD
Call groups allow to define a specific order of ringing for a phone call directed to a group of
extensions. According to a priority sequence, one or more extensions will ring; such sequence
ends when one of the extensions picks up the call or when ringing timeout is reached.
ACD manage incoming phone calls in a different way. Such calls are put on hold and
forwarded (according to the arrival order) to the queue members; user can define ringing
policy (simultaneous, round-robin, etc…).
Figure 3.19 PBX  Call Groups
In the tab ‘Groups’ it is shown a list of configured groups and ACD; ‘Incoming group’ appears
by default, it cannot be removed but only customized.
Other call groups and ACD can be enabled or deleted by selecting the relative checkbox;
clicking ‘Save’ button changes are applied.
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To modify an already existing call group or ACD, user must click on the pencil icon (column
‘Edit’). New call groups and ACD can be added by clicking on the specific link.
3.2.1 Call Groups
Call groups can be reached through a specific IVR selection or a Direct Inward Dialing (DID)
configured on a gateway.
Figure 3.20 Call Group editing screen
Each call group can include an arbitrary number of extensions; such extensions must be
provided with a priority level. When all extensions with same priority level are busy (or
ringing timeout is reached), PBX proceeds to forward the incoming call to a lower priority
level. During this process, caller hears the dial tone or a hold on music (if configured).
Whether no lower priority level extension is configured, the incoming call can be forwarded
to a voice mailbox.
3.2.1.1
Configuration
In the ‘Configuration’ section following settings can be edited:

Call Group name: alphanumeric characters admitted

Per priority ring duration: time interval (expressed in seconds) after which incoming
calls are forwarded to lower priority level extensions, or, possibly, to a failover
action.

Members: extensions included into the call group and related priority level. Up to 10
priority levels are admitted.
3.2.1.2
Courtesy Messages
In the ‘Courtesy Messages’ section user can:
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Rev.: 15-02-2015

Define which audio file must be played in case of all extensions with lowest priority
level are busy. If no audio file is configured, default audio message is "At the moment
all operators are busy."

Define which audio file must be played in case of all extensions with lowest priority
level do not answer within the ringing timout. If no audio file is configured, default
audio message is "At the moment operators are not available."
3.2.1.3
Failover action for unanswered and busy calls or unavailable extensions
In case of unanswered and busy calls or unavailable extensions, available failover actions are
those described in 3.1.3.8.
3.2.1.4
Call Group Time Check
Figure 3.21 Call Group Time Check
In this section a time check for each specific group can be configured. For further details on
time check configuration please see 3.6.1.
3.2.1.5
Call Recording
Figure 3.22 Enabling Call Recording for Call Groups
In this section user can enable call recording for a group incoming calls (both unconditionally
or on request, typing *1 to start and stop recording).
3.2.2 Automatic Call Distribuition
Incoming calls addressed to call ACD are managed according to FIFO policy (First In First Out).
Each ACD can include an arbitrary number of extensions.
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ACD configuration screen is organized similary to call groups’ one, small differences concern
the ‘Configuration’ section.
Figure 3.23 ACD editing screen
In the ‘Configuration’ section following settings can be edited:

ACD Name: alphanumeric characters admitted

Maximum waiting time: maximum time (in seconds) within the pending call must be
served. When the call is not served within this time, a failover action (defined in the
‘Failback for unanswered call for busy or unavailable’) is performed.

Maximum number of callers allowed in the ACD: maximum number of calls the ACD
can handle simultaneously (0 = unlimited).

ACD priority: a priority level can be assigned to the ACD. Whether two, or more,
concurrent phone calls are addressed to two, or more, different ACD with shared
members, the highest priority ACD will be served first (higher value = higher priority).
Within ‘Configuration’, two more frames must be configured: ‘Members’ and ‘Callers’.
Figure 3.24 ACD editing, Members
In ‘Members’ frame user can add/delete ACD members. In addition, it is possible to set
several parameters valid for all ACD members:

Ring strategy: different ring strategies are availabe (ring all, linear, least recent,
fewest calls, random, round-robin)
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
Per member ring duration: time interval (in seconds) defining how long each
member’s device will ring. If timeout expires, calls can be forwarded again to that
ACD member after a waiting time (corresponding to the following value).

After all members time out, retry after: time interval (in seconds) that must be
between two ringing cycles.

Rest interval: time interval (in seconds) that must be between last completed call
and the next one addressed to that ACD member.

Play this message to the member who picks up the call: user can define which audio
message must be played to the member who answers the call.

Report caller holding time: if selected, PBX notifies to the member who answers the
call how long the caller has been waiting.

Forward calls also to busy members: if selected, incoming calls can be forwarded also
to busy ACD members.
Figure 3.25 ACD editing, Callers
In ‘Callers’ frame it is possible to configure ACD policy towards the caller; user can define:

Which welcome audio message must be played when a caller enters the ACD.

What to do after the welcome message or, if no message is configured, when a caller
enters the ACD. User can choose between playing the dial tone or a hold on music.
Further options can be configured in ‘Position announcements’ and ‘Custom announcements’
frames.
Position announcements:

Periodically announce position in ACD: if enabled, caller will hear a message notifying
his position in the queue.

Announce the estimated waiting time: if enabled, a message will announce expected
waiting time. Such message can be played once, when the caller enters in the ACD, or
several times.

Announce frequency: time interval (in seconds), indicates how often previous
messages must be notified to the caller.
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Custom announcements:

Play the file: if this checkbox is selected, user can configure which audio message
must be played.

Each: user can specify how often (in seconds) such message must be played.
In addition, a DTMF code can be defined; typing this code the extension will manually start
the play of a prerecorded message.
Other features (failover action, time check and call recording) can be configured as shown in
‘Call Group’ paragraph, for more details please see 3.2.1.
3.3 Gateway
In the ‘Gateway’ tab user can configure all incoming/outbound lines. KalliopePBX supports
both, physical gateways (needed to connect local phone network to external analogical lines)
and VoIP Terminations and Trunks, using standard SIP. Multiple gateways and VoIP
Terminations and Trunks can be configured simultaneously on the PBX, each user can decide
which outbound line intends to use.
Usually, when a user dials an external number (prefixed by 0, line reservation code),
KalliopePBX selects the routing line according to LCR (Least Cost Routing) Class assigned to
the user and previously defined in PBX  Outbound Routing. Line reservation code can be,
optionally, disabled in PBX  Outbound Routing.
‘Gateway’ configuration tab is divided into two forms: ‘Options’ (to set global paremeteres)
and ‘Gateways/VoIP Terminations and Trunks’.
3.3.1 Options
Figure 3.26 Global options for outbound calls
In this form following options are available:

Enable incoming calls from unknown peers: checkbox disabled by default. When
KalliopePBX receives a phone call, it checks whether SIP From User Part matches with
a local extension, a registered gateway or a configured Trunk VoIP. PBX automatically
accepts such calls, no authetication is needed. Otherwise, if a call comes from an
unknown domain, PBX rejects it. When this option is enabled, incoming calls from
unknown domains are accepted and forwarded to the Internal Numbering Plan since
specific routing cannot be applied to such calls.
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Note: please pay attention, when this function is enabled, external users
can access internal Numbering Plan and place unauthorized outbound calls.

Enable Ringing Courtesy Tone: if enabled, caller will hear a dial tone while the
gateway is still routing outbound call, otherwise, no tone will be heard during the
routing process.
Note: at SIP protocol level, this option implies that a ‘180 Ringing’ message
is sent to the caller even if such signal has not actually arrived.
For external calls, it may create confusion in case of a busy addressee, in
fact PBX will previously play a ringback tone and then a busy one.
Moreover, when this option is enabled, pre-call provider messages (such as
“user not available”) cannot be delivered to the caller. In fact, at SIP
level, these messages are played when a ‘183 Session Progress’ message is
sent.

Timeout for outbound calls: user can specify time interval (in seconds) thereafter PBX
considers an outbound call as ‘Not answered’.

Balancer Traffic Profiles: these traffic profiles must be assigned to outbound lines
used in a balancer. A balancer is a virtual gateway used to distribute outbound calls
among several outbound physical lines in order to balance their use and set a
utilization limit for each line (expressed in seconds for a specific period).
By clicking on ‘Add new traffic profile’ a new profile can be created, user needs to
set following parameters:

Profile Name

Limit (seconds per period)

Type (seconds per month)

Additional seconds when configured limit is reached.
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3.3.2 Gateways/VoIP Terminations and Trunks
Figure 3.27 Gatewats/VoIP Terminations and Trunks
This form is organized into three sections: Gateways, VoIP Trunks and Terminations, and
Balancers. For each section, a recapitulatory list is available.

Gateways: such devices can have a static configuration (an IP address must be
assigned) or a dynamic configuration (the gateway must be registered on the PBX with
SIP credentials).

o
Registration State (if dynamic) or SIP accessibility (if static)
o
Round-Trip Time (RTT) for a SIP message between PBX and Gateway
o
Gateway ID
o
Gateway Name
o
Gateway Selection Prefix
o
Maximum concurrent calls
VoIP Terminations and Trunks: PBX registers on VoIP providers using one or more
identities. In both cases outbound lines are grouped for registration domain.
o
For each domain, following information is shown:

Domain SIP accessibility

Round-Trip Time (RTT) for a SIP message between PBX and VoIP
provider

VoIP domain/server name (KalliopePBX supports
both domain
configurations, DNS SRV query and hostname/IP address)

o
Domain Name
For VoIP Terminations and Trunks, following information is shown:
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Rev.: 15-02-2015

Account Name

Account Registration State (grey LED = registration is not operative)

Type: Trunk or Termination

Account Username (usually different from the authentication name, it
may represent the geographical number associated to such account)

Outbound Line Reservation Prefix

Maximum concurrent calls
There is a main difference between Terminations and Trunks: to the former only one phone
number can be associated for each authentication login, while, to the latter more phone
numbers (usually with a common prefix) can share the same credentials.
For each item in the list, checkboxes to perform general actions are available: ‘Enable’
checkbox allows to enable/disable a specific outbound line, ‘Delete’ checkbox allows a
multiple lines selection in order to delete them quickly from the PBX. Changes become
operative clicking on the ‘Save’ button.
If a checkbox is not editable, it is shown as faded; it means that such gateway is configured
as default outbound line or assigned to a LCR rule. Moving the mouse over the faded
checkbox, LCR rule information is shown. In order to disable/delete one of these outbound
lines, it must be previously removed from the LCR rule configuration.
Gateways, VoIP Terminations and Trunks settings can be modified by clicking on the pencil
icon (column ‘Edit’) or directly on the item’s name.
New Gateway, VoIP Terminations or Trunks can be added to the list by clicking on the
dedicated buttons.
For VoIP Terminations
and Trunks
user can define on which domain
the VoIP
trunk/termination must be created; such domain can be created, or selected from a set of
preconfigured ones. For each domain, multiple trunks or terminations can be defined. Each
trunk or termination can be used as an independent line for outbound calls; in case of
incoming calls, instead, routing rules are defined in the ‘VoIP Domain’ section.
3.3.2.1
Add/Edit Gateway
‘Add/Edit Gateway’ tab in organized into five different sections: ‘Account’, ‘Configuration’,
‘Outbound Calls Policy’, ‘Direct Inward Dialing (DID)’ and ‘Codec Selection’.
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Figure 3.28 Add/Editing Gateway screen
In the ‘Account’ section user can define following parameters:

Gateway ID: used to identify the gateway in the PBX system. Only alphanumeric
values admitted (no special characters), it must be unique (no other gateways, VoIP
trunks or terminations, or extensions can have the same ID).

Host: define whether the gateway is static or dynamic.
o
In case of static hosts, no gateway registration on the PBX is needed. IP
address and port must be defined to identify and accept phone calls coming
from such gateway.
o
In case of dynamic hosts, it is needed a gateway registration on the PBX using
SIP
credentials
(username/password).
Password
field
only
admits
alphanumeric values, between 3 and 30 characters. In such case, user can
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Rev.: 15-02-2015
also decide if incoming calls need to be authenticated with registration
credentials.

On incoming calls, ignore the source port when recognizing the origin of the call: user
can enable peer recognition based only on IP address (without any check on the
source port). This option is useful when a firewall modifies SIP messages source port.

Disable: this checkbox allows to temporarily disable the gateway; specific gateway
configuration is preserved.
In the ‘Configuration’ section user can define following parameters:

Gateway Name: the name that will be displayed in selection lists and CDRs.

Gateway Selection Prefix: (optional) it is a prefix used to explicitly select an outgoing
line (bypassing LCR classes). Such prefix is automatically removed by the PBX when
forwarding the call to the gateway.

Backup Gateway: user can specify a backup gateway to use when a congestion (busy
lines) or a fault occur on the gateway. In order to prevent cyclic call forwarding, it is
important to avoid loops when defining the backup gateway.

Maximum concurrent calls/Number of available lines: user can define how many
concurrent calls a gateway can handle. When such limit is reached, a further call will
not be allowed and a message will notify the user that all outgoing lines are busy.
Note: when the maximum concurrent calls limit is reached, PBX does not
forward further calls to such gateway and behave as if a congestion
occurred.
When no limit is defined, if gateway cannot handle an additional call (e.g.
2 concurrent calls on a gateway with a single ISDN BRI interface), such
gateway refuses the call with a Failure Message (tipically a SIP ‘503 Intenal
Server Error’ message).
In the ‘Outbound Calls Policy’ section user can define the following parameters:

Add this prefix to called number: user can define a prefix to add to the outbound
number when forwarding a call to the gateway.

DTMF send mode: DTMF sending policy when forwarding to such gateway (admitted
options are RFC2833, SIP Info, Inband).

Use this traffic profile for balancer: user can define which traffic profile must be
applied to the gateway when a balancer is configured.

Trust RPID/PAI: if enabled, the Remote Party or P-Asserted-Identity header must be
trusted for calls with that specific gateway.

Add P-Asserted-Identity header to outbound calls: P-Asserted-Identity header allows
to display the caller identity on outbound calls. It is suggested not to enable such
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option since outbound calls could be refused when addressed devices do not support
it.

Enable send Call-Info: this checkbox allows to add the Call-Info field to SIP INVITE
Header in the following format:
Call-Info: <extension> ;purpose=info
In the ‘Direct Inward Dialing (DID)’ section it is displayed a list of actions to perform in case
of incoming calls addressed to phone numbers associated to such gateway. By clicking on the
‘Edit’ button user can open DID configuration screen.
Figure 3.29 PBX  Gateways  Add/Edit Gateway, DID configuration screen
In this screen user can define, or modify, which action must be applied to incoming calls
addressed to the gateway. A specific action can be performed depending on the called
number12 (Called Party Number -CDPN).
For each number (with exact or prefix match) Callback service13 can be enabled; it must be
also specified the outbound gateway to use for such service. More information on Callback
service will be provided in ‘Incoming Calls Policy’ paragraph.
Whether Callback service is disabled, or the incoming call number is not included into the
callback list, such call will be forwarded to one of the following destinations:

Default incoming destination (as configured in PBX  Incoming Calls Policy)

Internal Numbering Plan
o
Keeping the called number
o
Changing the called number to a specific value
o
Requiring the caller a selection

Local (or remote) extension

Callgroup or ACD (taking into consideration the relative opening time check)

IVR menu/submenu

Opering time check

Audioconference service (optionally specifying the room)

DISA service
12
In case of gateways with multiple ISDN port in multi-number mode or ISDN PRI lines with
Short Numbers option enabled.
13
Callback service checks if caller number is included in a list defined in PBX  Incoming
Calls Policy. In case of positive match, PBX aborts the call, recalls the number and connects it
to the desired destination using the configured gateway.
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
Ignore the call (incoming call keeps on ringing, but PBX does not route it to any
destination)

Refuse the call (incoming call is rejected).
In the ‘Codec Selection’ section it is possible to define a codec list valid for the specific
gateway, overlooking the global settings configured in PBX  Features. By selecting the
checkbox ‘Enable custom codec selection’ user can define audio and video codec (supported
codec are: PCM a-law, PCM u-law, G.729, G.722 and GSM for the audio, and H.261, h.263,
H.263+ and H.264 for the video).
3.3.2.2
VoIP Terminations
A new VoIP Termination can be created by clicking on the ‘Add new VoIP termination’ button.
Each new termination must be associated to a VoIP domain (a preset or a new one) by
selecting it from a drop-down menu.
Figure 3.30 PBX  Gateway  Add new VoIP termination, VoIP domain selection
As for the gateways, when adding a new VoIP termination the specific tab is organized into
several sections: ‘VoIP Domain’, ‘Account’, ‘Configuration’, ‘Outbound Calls Policy’, ‘Direct
Inward Dialing (DID)’ and ‘Codec Selection’.
In the ‘VoIP Domain’ section user can define the following parameters (only for new
domains):

Name: VoIP Domain ID

Domain/Server (port): server name (or domain, that will be converted to server name
through a DNS SRV query) on which the new termination must be authenticated. Port
must be specified only for server name (not for domain) and only if different from the
default one (5060)14.
Note: this section is editable only when creating a new VoIP domain. When a termination is
assigned to an already existing domain such section is displayed in read-only mode.
In addition, after initial configuration, it is possible to access domain configuration tab in
order to define incoming calls policy (for example user can change the DTMF tones receive
mode or supported codec).
14
If user inserts the default value 5060, it will be automatically removed when saving.
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Figure 3.31 PBX  Gateway  Add new VoIP termination, Account
In the ‘Account’ section user can define the following parameters:

VoIP Number/Account: alphanumeric characters admitted. Usually corresponds to the
geographical number assigned by the VoIP provider; such number determines the DID
incoming calls routing. Outbound calls from this termination will use this value as
caller ID.

Authentication Name: login used by the PBX to authenticate registrations and
outgoing calls. It is not a strictly numerical value; when coinciding with the account
name it can be omitted.

Password: password used by the PBX to authenticate registrations and outgoing calls.
This entry admits alphanumerical values, between 3 and 30 characters.

Enable SIP registration: when PBX registration on the VoIP domain is not required this
option can be disabled in order to prevent PBX to send periodic REGISTER messages to
the domain.

Disable: this checkbox allows to temporarily disable the termination, specific
configuration is preserved.
‘Configuration’ section is comparable to the corresponding one when adding a new gateway.
Figure 3.32 PBX  Gateway  Add new VoIP Termination, Configuration
‘Outboud Calls Policy’ section includes all the features available in the corresponding
gateway section. In addition, user can modify source domain in the SIP URI "From:" header for
outbound calls from this termination. By default, PBX uses termination registration domain,
but, user can specify a different domain or IP address.
Figure 3.33 PBX  Gateway  Add new VoIP Termination, Outbound Calls Policy
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Figure 3.34 PBX  Gateway  Add new VoIP Termination, DID configuration screen
As for gateways, this section includes Callback service checkbox and gateway selection for
such service.
In the ‘Codec Selection’ section user can define a codec list associated to the specific
termination, overlooking the global settings configured in PBX  Features.
Note: codec configuration related to specific VoIP terminations is only
valid for outbound calls. Incoming calls codec selection depends on the
VoIP domain configuration and it is valid for every termination associated
to such domain.
By clicking on a domain ID, it will be displayed a summary screen for such domain (including
the terminations defined on it).
Figure 3.35 VoIP Domain summary screen
In such page user can modify domain name, server and port, but also:

set the maximum concurrent calls admitted on this domain

set the DTMF send mode for incoming calls
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
ignore the source port when identifying the origin of the call

extract the called number from the header “To:” instead of the “Request-URI”, which
is the default behaviour.
In the ‘Direct Inward Dialing (DID)’ section it is shown a list of DID rules related to the
terminations configured on such domain. User can modify the incoming calls policy for each
specific termination by clicking on the relative pencil icon.
User can add DID rules for additional numbers by clicking on the button ‘Edit’. In case of
multiple numbers assigned to a single extension, incoming calls routing can be managed
differently for primary or secondary numbers. In case of outbound calls, displayed number
will always be the primary one.
In the ‘Codec Selection’ section user can define a codec list valid for the incoming calls
addressed to that specific domain, overlooking the global settings configured in PBX 
Features.
3.3.2.3
VoIP Trunks
In case of VoIP Trunks, it is provided a set of geographical numbers, mapped on PBX specific
extensions. General configuration is similar to the VoIP Terminations one, except for: number
masking for outbound calls, DID rules, remote numbering range (a set of numerical selections
that becomes part of the PBX Internal Numbering Plan, allowing automatic routing through
the trunk).
As for VoIP Terminations, the PBX registers and logins on the provider platform. When PBX
registration is not required, it is possible to disable it by selecting the dedicated checkbox. In
case of VoIP Trunks, authentication credentials consist in the pair username/password,
editable in ‘Account’ section.
The main difference, with respect to VoIP Terminations, consists in the possibility to modify
the outbound displayed numbers according to a mapping of the extension:

Extension numbers remain unchanged

Caller numbers are changed to a predetermined value

Caller numbers are obtained: removing some digits from the top of the caller number
and adding a prefix (for example, if user configure to remove one digit and add the
prefix 050999, the call made from 123 extension will be displayed on the Trunk as
05099923).
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Figure 3.36 PBX  Gateway  Add new Trunk, Outbound Calls Policy
Note: from firmware release 3.7.7, trunk outbound calls (after an ENUM
query) display the Caller ID according to the rules of the trunk (and no
longer according to the settings of the panel "ENUM", which has been
deleted).
With respect to incoming calls routing, DID rules must be set in the VoIP domain configuration
page. By default, incoming calls routing forwards them to the Internal Numbering Plan.
From firware release 3.8.0 it is possible to define remote numbering ranges, that is, sets of
numbers reachable through the trunk and considered by the PBX as normal extensions. Such
remote numbers are managed in a very similar way to the local extensions, they can: access
Internal Numbering Plan, use KalliopePBX phone services or reserve an outbound line
(according to the relative LCR class). In addition, remote numbers are included into the
Internal Numbering Plan, so that other extensions (both local or remote) can directly contact
them to their extension number (no routing prefix is needed).
Figure 3.37 PBX  Gateway  Add new Trunk, Remote numbering range
On the trunk in Figure 3.37 the following remote numbering range is configured:

extension 620, not enabled to place outbound calls from KalliopePBX

extensions from 610 to 615 (included), outbound calls are managed according to the
LCR class ‘default’

numbers with prefix 60 (e.g., 600, 601, ..., 609, 6000, 6001, ..., 6099, etc.).
To add, modify, delete remote numbering ranges or to assign them LCR classes, user can
access the configuration page by clicking on the ‘Edit’ button. In case of overlap between
different ranges within the same Trunk, PBX performs, in order, a match on: the exact
numbers, the range and the prefixes.
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Note: in order to define a remote extension, it must belong to a numbering
range defined for its corresponding trunk. When the remote extension is
defined, it is possible to overwrite its previous LCR class.
3.3.2.4
Balancers
Balancers are virtual gateways used to distribute outbound calls among several outbound
physical lines in order to balance their use and set a utilization limit for each line.
Figure 3.38 PBX  Gateway  Add New Balancer
In order to configure a Balancer user must define:

Name

Selection Prefix (as for physical gateways)

A backup gateway to use in case of congestion (or when traffic limit is reached) for
all the gateways associated with the balancer

One or more gateways associated to a traffic profile.
Balancers will appear in the gateways selection lists when configuring them, but, they can
also be explicitly selected by typing their selection prefix.
In case of outbound calls addressed to a Balancer, it checks which gateway has more residual
traffic15 and automatically forwards the call to such gateway. Whether the selected gateway
is busy or not available, Balancer chooses the gateway, from those available, with more
residual traffic. If all gateways have reached their traffic limit, Balancer chooses a gateway
configured with an additional traffic amount. When all gateways are busy, not available or
with no traffic left, outbound calls are forwarded to the configured backup gateway.
Balancer only functions with outbound calls, incoming calls are managed by physical
gateways.
15
Residual traffic is calculated analysing current month CDR.
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3.4 IVR
Figure 3.39 PBX  IVR
IVR (Interactive Voice Responder) allows the administrator to configure a vocal menu,
browseable thanks to the phone keys, dedicated to external incoming calls. Thanks to IVR,
incoming calls can be forwarded towards several destinations, previously configured in this
tab.
Figure 3.40 IVR Configuration
Configuration process consists in associating a phone key to an action available from a dropdown menu, and defining an audio message to guide the caller. It is also possible to create
submenus to better organize the IVR structure. If the caller presses an invalid key, the audio
message will be played again. Administrator can specify a selection timeout; when such
timeout is reached, one of the following predefined actions is perfomed:

Repetition of menu message (if caller’s device cannot send DTMF tones he will be
blocked in this menu)
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
Back to the main menu

Forward the call to a call group or an ACD.
Available actions for each phone key are:

Forward the call to an IVR menu

Forward the call to a call group or an ACD defined on the PBX

Forward the call to a specific extension chosen by the caller via phone keys (an audio
message will guide the user)

Forwar the call to a predefined extension

Forward the call to an opening time check

Forward the call to an audioconference room (both automatically or giving the user
the possibility to choose the room)

Forward the call to the DISA service

Hang up the call, optionally an explicative audio message can be played.
3.5 Features
The ‘Features’ tab allows the user to edit:

KalliopePBX on-call services

SIP settings

Sender e-mail address used for e-mail notifications and voice messages

Default audio/video codec.
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Figure 3.41 PBX  Features
3.5.1 On-call Services
3.5.1.1
Blind Transfer (#4)
Such function allows to transfer a call between A and B to C. The transferer will press #4 and
a voice will ask to type the extension number he wants to forward the call to. During this
process a hold-on tune will be played to the transferred user. C’s device will ring and, when
he will answer, he will be direclty connected to the transferred user. Whether C is busy or
not available, the call will return to the transferer and the number on the display will be
preceeded by a ‘R’.
3.5.1.2
Attended Transfer (*4)
Such function allows the transferer (after typing *4 followed by the extension number he
wants to forward the call to and #) to talk with C before performing the transfer. Only when
the transferer hangs up, C is connected to the trasferred user. If C does not accept the
transfer, the call between A and B will be re-established. Transferer can restore the call with
the trasferred user (while C phone is ringing) by typing *0.
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3.5.1.3
Call Parking (#8)
Such function allows to put on-hold a call and recover it from any phone. Typing #8 during a
call, the caller is parked and an audio message notifies the user which number must press to
recover it. PBX has an editable number of parking slots that can be defined in the ‘Numbering
Plan’ tab.
After 45 seconds a call has been parked, it is forwarded back to the extension that had
parked it.
3.5.1.4
Fast Transfer (**)
Such function allows a blind transfer between an extension and the associated mobile
number, and vice versa. It is available only for those extensions with a configured mobile
number.
3.5.2 SIP Settings
‘SIP Settings’ section includes:

Enable pedantic control of SIP messages: enabled by default, this option forces
KalliopePBX to perform a sintactic check on the received SIP signaling. Disabling this
function may solve some compatibility issues with UAs not strictly SIP compliant

Enable SIP transport over TCP: disabled by default, it allows to support SIP over TCP.
The preferred transport method is UDP, but, whether a UA registers on PBX using
TCP, then all the signaling towards this UA will be performed using TCP. When two
peers use different transport protocols, KalliopePBX performs the adaptation between
TCP and UDP.

“Activate VoIP services on the port”: UDP port (and possibly TCP). Default value is
the standard 5060.

“DSCP xxx configuration”: DSCP values to tag the signal outgoing from Kalliope, in
order to announce the correct treatement priority required to the network.
3.5.3 Voice Mailbox
This field allows to modify the e-mail address used as sender for e-mail notifications.
3.5.4 Audio Codec Selection
KalliopePBX supports following audio codec: G.729, PCM u-law, PCM a-law, GSM, G.72616 and
G.722. In this section user can define which codec (and in which order) must be used.
Note: G.729 is licensed and, normally, it can be used ONLY in “passthrough” mode, that is, when both the endpoints of the call are using this
codec. Whether the PBX needs to handle a G.729 media flow (for example
to access the voice mailbox or to record an audio file) or one of the
endpoints does not use G.729, it is necessary to buy a license. Each license
is dedicated to the handling of a single G.729 media flow. For further
16
At 32 Kbps.
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details on licenses activation please see paragraph 2.11.
3.5.4.1
Video calls support
PBX supports following video codec: H.261, H.263, H.263+ and H.264. In this section user can
enable video calls support and define which codec (and in which order) must be used. Both
endpoints must share the same video codec in order to perform a successful video call.
3.5.4.2
Enable T.38 Support
KallioepPBX supports T.38 protocol for fax handling in pass-through mode, so both endpoints
must be provided with T.38 protocol support.
3.6 Incoming Calls Policy
In the ‘Incoming Calls Policy’ tab user can define how to behave when PBX receives incoming
calls not corresponding to DID rules configured on gateways or VoIP terminations (except for
VoIP trunks since, by default, they forward the call to the extension associated to the number
called).
Figure 3.42 PBX  Incoming Calls Policy
3.6.1 Opening Time Check
From firmware release 3.7.1 it is possible to manage multiple time checks. A list of
configured time checks is displayed and a new one can be added by clicking on the ‘Add New
Time Check’ button. The calendar icon allows to show a list of opening days and opening
hours related to that specific time check.
By clicking on the name of the time check (or on the pencil icon) the configuration page is
opened.
After playing an optional audio message, time check proceeds with a comparation between
current system time and user-defined time ranges. In particular, user can define:

Specific closing days, with optional annual repetition

Weekly opening time, associating to each day of the week different opening times (or
complete opening).
In order to modify closing days or opening times, user can click on the relative ‘Edit’ button.
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Figure 3.43 Time Check configuration screen
Figure 3.44 Closing days configuration screen
Closing days can be full (24 hours) or admit multiple opening times.
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Figure 3.45 Opening time configuration screen
Weekly timetable can be configured both as opening time or closing time, specifying one or
more time ranges for one or more days of the week.
Note: setting the timetable as normally closed, without defining any
opening range, the time check will always return a ‘Closed’ status.
For each status (closing day, weekly closing time, opening time) a different destination can
be defined. After an optional message, one of the following actions can be performed:

Hang up

Forward to a specific extension

Forward to a call group (with or without a specific time check)

Forward to an ACD (with or without a specific time check)

Forward to a voice mailbox

Forward to an IVR menu or submenu

Forward to an external number, specifying the gateway to use

Foward to another time check.
In case of closing time, the caller can also bypass time check by typing a backdoor code and
proceed as if it were an opening time.
3.6.2 Incoming Calls Policy
For those incoming calls that follow the default routing (that is, not forwarded to call groups,
ACD etc.) a welcome message can be defined. Afterwards, such calls are addressed to the
destination selected from the drop-down menu.
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In addition, it is possible to add a prefix to the caller ID; user can choose between ‘none’ and
‘0*’. If ‘0*’ is selected, such prefix will be added to the caller ID, in order to give the user the
possibility to recall it directly from the call history on his phone.
In the same section, user can enable and configure Callback and DISA services.
3.6.3 Callback Service
Figure 3.46 Callback service configuration
Callback service includes a list of external phone numbers enabled to be automically recalled
by the PBX; the switchboard rejects the incoming call and immediately recalls its number.
When this call is answered, the PBX will actuate as if it were a normal incoming call,
forwarding it to the desired destination. In addition, user can specify which gateway must be
used for the recall.
Callback service can be enabled by selecting the specific checkbox; the list of enabled
numbers (and their relative destinations) can be edited by clicking on the button ‘Edit’.
Figure 3.47 Callback enabled numbers configuration screen
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3.6.4 DISA Service
Figure 3.48 DISA Service configuration screen
DISA service is accessible by local extensions by typing the default code 803, or by external
users through an IVR menu or typing a code (a voice message will ask the user to provide the
authentication credentials). If authentication is successful, the system will ask the user to
type the number to call; such call is equivalent to those originated from a local extension, its
routing policy correspond to the LCR class associated to the DISA user.
3.7 Numbering Plan
Numbering Plan defines internal routing policy for calls handled by KalliopePBX. Such plan can
be differentiated in case of:

calls originated from a PBX extension (both local or remote)

incoming calls originated from an external number
The call selection operates by checking the numbering plan rules until a match is found,
according to the following priorities:
1. Telephone services
2. Speed-dial (user defined or system wide)
3. User defined selections (based on an exact, a range or a prefix match), associated to
a call group, an ACD, an IVR menu etc. Number manipulation is also possible, before
forwarding it back to the numbering plan
4. Local extensions
5. Remote extensions (accessible via VoIP Trunks)
6. Outbound line reservation using a reservation code associated to a specific
gateway/Trunk/VoIP Termination
7. Outbound line reservation according to the LCR rules associated to that PBX user.
The ‘Services’ column allows the user to modify, enable or disable specific phone services.
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Figure 3.49 PBX  Internal Numbering Plan
Services configuration page can be accessed by clicking on the pencil icon.
Figure 3.50 PBX  Internal Numbering Plan, Services configuration screen
Here below an overview of available phone services.
3.7.1 Call-pickup Group
This function allows any extension to pickup an incoming call addressed to a different
extension belonging to the same group. In case of busy or unavailable extension, another user
can answer such call from his own device. Default call-pickup code is *9.
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3.7.2 Direct Pickup
In order to pickup an incoming call addressed to a specific extension, user can dial direct
pickup code (**) followed by the extension he wants to pickup. Such function is not bound to
call groups.
3.7.3 Call Pickup with Invitation
For further details about this function please see 3.8.3 firmware release changelog.
3.7.4 Callback Request
Whether the called extension is busy, caller may request a callback by typing, within 20
seconds, the specific code. Once the extension is available, PBX automatically calls back the
caller and puts him in contact with the called user. Caller may annul a requested callback by
typing the specific cancellation code.
3.7.5 Echo Service
Thanks to this service user can hear back his own voice, this may be useful to check the
correct functioning of the device.
3.7.6 Voice Mailbox
Each PBX user can access, within the network, his voice mailbox by typing the specific code
(by default 801). In order to allow users to access voice mailboxes from outside the network,
the specific checkbox must be selected in PBX  Numbering Plan.
User is asked to provide mailbox number and password; available actions (such as
listen/delete a message) can be performed thanks to the IVR menu.
In addition, user can modify the notification sender e-mail address.
3.7.7 Audioconference Service
By typing the specific service code, user will be asked to provide the number of the room he
wants to be connected to.
3.7.8 DISA Service
For a detailed description of such service please see paragraph 3.6.4.
3.7.9 Electronic Lock
A code must be specified in order to lock/unlock a specific extension.
3.7.10 Unconditional Forward
Such function allows the PBX to forward incoming calls addressed to a specific extension (but
not to call groups or ACD) to a different extension.
User can specify enable/disable code for this function. In order to enable unconditional
forward, user must type the relative code followed by the extension number to forward the
call to; to annul unconditional forward user must type, from his device, the cancellation
code.
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A BLF (Busy Lamp Field) is available for all BLF groups in order to display unconditional
forward status.
Such function admits call forward to external numbers.
3.7.11 Call Tagging (available with the optional Call Center module)
This function permits to tag each outbound call with a specific Account code, also showed in
the CDR. The Account code is assigned to a call using a service prefix that must be configured
in the Numbering plan. For example, if the service prefix configured is *1, to tag a call with
the Account code 987 the user must dial *1 987 <destination>, where <destination> is the
number to call (including the 0 prefix if configured).
The Account code can also be set through the API Rest “/click2call”, inserting in the request
the tag <accountcode> (please see the API Rest documentation available on
www.kalliopepbx.com for more information).
3.7.12 Spy call supervision
Prefix that enables Supervisors to spy target Operators during calls.
3.7.13 Dual Ring
When configured and enabled, incoming calls are forwarded both, to the extension and to the
associated mobile number.
A BLF (Busy Lamp Field) is available for all BLF groups in order to display dual ring status.
3.7.14 Call Parking
User can configure parking slots range. A BLF (Busy Lamp Field) is available for all BLF groups
in order to display call parking status.
Parking code allows to park an ongoing call into a slot (default code is #8).
The following section (‘Custom numbering plan rules’) allows the user to configure custom
routing policy for incoming calls. An incoming call can be forwarded to: a call group or ACD,
an opening time check, an IVR menu, an audioconference room, a specific extension or a
custom selection to set in the Numbering Plan.
Figure 3.51 Internal Numbering Plan, Custom Numbering Plan rules
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Such actions can be performed identifying incoming calls on the base of: extact match, range
match or prefix match.
Custom rules can be created, deleted or modified by clicking on the pencil icon. Please note:
custom rules are automatically ordered favoring exact match, range match and, eventually,
prefix match.
Figure 3.52 Custom Numbering Plan rules configuration screen
Remaining sections concern local extensions, remote extensions, outbound calls with explicit
line reservation and outbound calls with line reservation using LCR rules.
Figure 3.53 PBX  Numbering Plan, Local and Remote Extensions and Outbound Routing
Whether the selection does not match any rule it is performed a “catch-all” action. By
default, such action consists in refusing the call: with firmware versions lower then 3.12 an
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audio file notifies the user that the selection is not valid, starting with firmware version 3.12
the “catch-all” action can be customized and a specific destination can be set. Whether the
configured action is “Refuse” no audio file is reproduced and the error 404 (Not Found) is
notified to the caller.
3.8 Outbound Routing
Figure 3.54 PBX  Outbound routing
In this tab it is possible to define outbound routing policy in case of users dial 0 (if enabled)
or 0*, that is whether they do not explicitely select a gateway.
‘Outbound Routing’ tab is organized into different sections as shown below.
‘Outbound Routing Options’ section includes:

International prefix field

Country Code field (intended as where the PBX is located)

Omit 0 for outbound calls checkbox

ENUM Networks field

LCR rules for call forward to external numbers.
‘LCR Classes’ section allows the user to assign a specific LCR class to an extension or to a
range of extensions. All LCR classes have a default final rule that can be ‘Allow’ or ‘Block’.
Outbound routing using LCR rules can be preceded (if enabled) by an ENUM check on a set of
defined domains. If one of such domains returns a valid URI SIP, then the call is routed via
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VoIP. Whether the call fails (but not in case of busy line), such call is routed using the related
LCR class.
Each LCR class is defined by an ordered set of LCR rules, when a call is placed PBX applies a
first-match policy.
‘LCR Rules’ section allows the user to assign a set of destinations (prefixes, exact numbers) to
each outbound gateway (if not, then the final rule is ‘block’). Thanks to different
combinations of preset rules, user can configure a specific LCR class for each PBX extension.
Here below is an example of possible LCR rules assinged to a LCR class:

‘Mobile phones’ rule: prefix 3  Gateway ‘A’

‘International calls’ allow rule: prefix 00  Gateway ‘B’

‘International calls’ block rule: prefix 00  Block

‘High fares’ rule: prefixes 899, 199, etc…  Block

‘No restriction’ rule: any number  Gateway ‘C’
LCR classes can be assigned in order to manage extensions’ authorizations, for example:


LCR Class A:
◦
‘Mobile phones’ rule
◦
‘International calls’ block rule
◦
‘High fares’ rule
◦
‘No restriction’ rule
LCR Class B:
◦
‘Mobile phones’ rule
◦
‘International calls’ block rule
◦
‘No restriction’ rule
Differently from LCR Class A users, LCR Class B users are allowed to call ‘High fares’ numbers.
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Chapter 4 -
PBX Applications
4.1 External API
This panel permits to configure phone calls routing basing on the response of an external API.
Such API is invoked providing a series of numerical parameters that can be requested to the
user thanks to a set of pre-recorded prompt messages.
Figure 4.1 New external API configuration, Global settings
The API must be served by a web server external to KalliopePBX. The configuration process
includes the following parameteres:

API’s URL: it can be an http o https URL (without server certificate validation)

Authetication:
it
can
be
NONE
(no
authentication)
or
BASIC
(Basic
http
Atuhentication); in this last case, it is needed to provide authentication credentials
(username and password)

Request type: API invocation can be GET or POST; in the first case the contingent
parameters are provided through the request URL, while in the second case it is
possible to specify the request content format and the relative Content-Type.
In both cases (GET and POST) API’s parameters are transmitted through placeholders included
in the URL or in the POST content. Available placeholders are:

%CALLER_NUM%: caller number

%PARAM1%, …, %PARAMnn%: “nn” parameters required to the caller through an IVR
built as described here below.
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For example, a GET the URL request could be built as follows:
http://www.myserver.com/api?arg1=%CALLER_NUM%&arg2=%PARAM1%&arg3=%PARAM2%
while for a POST request-type the content could be:
<?xml version="1.0"?>
<parameters>
<caller>%CALLER_NUM%</caller>
<param_01>%PARAM1%</param_01>
<param_02>%PARAM2%</param_02>
</parameters>
For each numerical parameter required to the caller it must be configured a vocal message.
Optionally, it is possible to establish the maximum number of digits admitted; if no value is
set, the system considers the input over after a 5 seconds timeout or after the pression of the
# key. Whether no input is provided, the request is repeated three more times.
Figure 4.2 New external API configuration, Parameters acquisition
In addition, it is possible to enable the confirmation request by selecting the dedicated
checkbox; if enabled, the system asks the user to confirm the inserted digits by pressing the
key ‘1’. Whether the user does not confirm the parameters, the insertion request is repeated.
Once the parameters are correctly provided, the configured external API is invoked. The
system expects a “200 OK” response and its body must contain an XML text as the following
one.
<?xml version="1..0"?>
<response>
<message>
<elem>digit:1</elem>
<elem>number:200</elem>
<elem>alpha:c234</elem>
<elem>audio:custom/misc/pluto</elem>
<elem>number:201</elem>
<elem>digit:123</elem>
<elem>audio:custom/misc/pippo</elem>
</message>
<value>105</value>
</response>
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Figure 4.3 New external API configuration, Actions configuration
The <value> tag indicates the API answer depending on which several actions can be
perfomed (starting with firmware release 3.10 this value can be 0-9 and not only 0 or 1).
KalliopePBX can play an introduction message to the caller (defined in the configuration web
GUI) and series of other messages connected basing on the <elem> tags sequence present in
the <message> answer. The <elem> tags admit the following values:

digit: plays single digits one-by-one (multiple digits numbers are admitted, but the
digits will be played one-by-one)

number: plays the whole number

alpha: admits digits and letters that are played one-by-one

audio: path of an audio file (stored into the PBX) to play
One of the available actions consists in the possibility to forward the call to the Numbering
Plan, using as selection the API answer value. Finally, it is also possible to set an action to
perform in case of API invocation error.
4.2 MeetMe Conference
MeetMe Conference tab allows the user to create and configure audioconference rooms. In
the main screen it is displayed a list of configured rooms, including names and numbers.
Two new tabs have been added from firmware version 3.12: “MeetMe Rooms Operation” and
“MeetMe Rooms MOH”.
The first one permits to define the rooms’ access and configuration permissions: in particular,
the admin can assign the control of a room (private room) to an extension that can manage
such room from its own KalliopePBX page.
The room owner (system admin or an extension) can: define an access PIN and an
administration PIN, enable the function “Announce”, enable the function “Automatically
close the room when the last admin user leaves”. When the “Announce” checkbox is
selected, the system asks the new participants to pronounce their name so to announce when
someone enters or exits the room.
Moreover, the room admin can configure an audio file that is used as hold-on music when
there is only a participant in the room.
To enter a specific audioconference room, user must press (after typing 802, the code
associated by default to this service) the room number followed by #. If needed, user is asked
to provide a password and, eventually, is forwarded to the room.
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An available shortcut is to type directly the number <SERVICE_CODE>NNNN*MMMM, where
(NNNN) is the room number, and (MMMM) the relative password. Whether the room is not
protected with an access password user must type <SERVICE_CODE>NNNN.
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Phonebook
Chapter 5 -
5.1 Phonebook
Phonebook tab allows the user to perform several actions:

Browse the phonebooks (local, personal and shared)

Configure LDAP parameters in order to download remote phonebooks

Define publication policy for KalliopePBX extensions on the local LDAP server.
Phonebook includes a list of phone numbers (PBX extensions or external numbers) easily
accessible and available for click-to-call service.
Note: click-to-call service is available only for PBX users with an associated
extension.
5.1.1 ‘Admin’ login
Accessing the PBX web interface at admin level, it will be displayed the following screen.
Figure 5.1 Phonebook
Phonebook tab is organized into different sections:

Extensions Phonebook: it includes every account configured on KalliopePBX (such
contacts are visible to all users).
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
Shared Phonebook: it includes every account created by KalliopePBX administrator.
Each user can decide to add or not his account to shared Phonebook (such contacts
are visible to all users).

LDAP Phonebook: it includes contacts downloaded from remote LDAP directories
(configured in the specific panel).

Search all: in this section user can perform a research on all available phonebooks.
In addition, each user is provided with a personal and editable phonebook:

Personal Phonebook: each user can decide which contacts must be added to his
phonebook (such contacts are visible only to that specific user). To access personal
phonebook, user must login at user level.
Figure 5.2 Shared Phonebook section
In ‘Shared Phonebook’ section administrator can:

Add new contacts (individually or importing a XLS/CSV file)

Export the contact list

Remove contacts

Enable/Disable the visualization of the contact ID instead of its number for incoming
calls (for internal calls, PBX always replaces caller ID with the value ‘Name Last
Name’ associated to the extension). The ‘Name Last Name’ value is researched,
firstly, in the private phonebook, then in the shared one and, eventually, in remote
phonebooks. If no entry is found, it will be displayed the caller's number.
Administrator can broswe, add, edit and remove contacts only from the System Phonebook.
In addition, each user can be associated with a speed-dial code (# followed by one or more
characters). Speed dial codes begininning with #0 are reserved to Personal Phonebook
contacts, so System Phonebook speed-dial codes must begin with # followed by one or more
characters different from 0.
Each user can perform the same actions with respect to his personal phonebooks.
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Figure 5.3 Add new contact to Personal/Shared Phonebook screen
Users and administrator can import phonebook contacts from XLS (Microsoft Excel worksheet
97/2000/2003) or CSV (Comma Separated Values) files.
In order to import contacts from a file, it must be clicked on the button ‘Import from CSV or
XLS’.
CSV/XLS files must include (not necessarily in the same orden):

GIVENNAME: first name (required)

SURNAME: last name (required)

O: Organization or Company

OU: Office or Unit

PHONE: phone number (required)

SPEEDDIAL: associated speed-dial, including the # (or #0) prefix

MAIL: e-mail address
In case of CSV files user can also define the separator character (typically ';' or ',').
Figure 5.4 Import from CSV/XLS screen
Clicking on the ‘Choose File’ button a file browser is displayed; user must select the file to
upload and click on ‘Upload’. In order to check for errors, a recapitulation of the uploading
contacts is shown.
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Whether one, or more, contacts are already present in the Phonebook, user can decide how
to proceed: already existing contacts can be updated (they will appear twice in the
Phonebook) or overwritten.
Figure 5.5 Contacts Import preview
5.1.2 ‘User’ login
Accessing the PBX web interface at user level a green phone icon is displayed next to each
contact; by clicking on such icon, user’s device starts ringing and it will be displayed ‘Web
call to xxx’ (where xxx is the called contact).
Figure 5.6 Personal Phonebook section
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Note: user login credentials correspond, initially, to the SIP account
credentials. Administrator can modify both, username and password for
each user. Each user can modify his credentials from the dedicated
section.
Whether the user is provided with ‘Shared Phonebook’ editing authorization, he will also have
the possibility to add, modify and remove contacts from such phonebook.
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5.2 Publish extensions on LDAP
Figure 5.7 Publish extensions on LDAP screen
Thanks to this page user can configure parameters required to publish extensions on
KalliopePBX LDAP server.
The LDAP tree can be built from an existing RDN; each entry can be based on a standard
template or on a customized one, obtained by editing the text area in the page. When
publishing, each field contained between the symbols %…% will be replaced by the actual
values.
The ‘Organization section’ frame contains the fixed part of the LDIF used to populate the
LDAP tree, while the ‘Person section’ frame shows the scheme used to publish each
extension.
The %PHONE% field is generated by applying the rules configured for each extension; by
default the extension number is modified by removing a configurable number of digits at the
beginning and adding a prefix.
LDAP Phonebook is automatically regenerated when KalliopePBX boots or whenever changes
are performed (and applied) to relevant fields.
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5.3 Publish Shared Phonebook on LDAP
It is also possible to publish Shared Phonebook on a different LDAP tree in order to make it
directly accessible from devices with a built-in LDAP client.
In such case LDIF structure is set by default, user can only modify the Root DN of the tree.
Default value is ‘dc = contacts, dc = kalliopepbx, dc = com’.
5.4 Import from LDAP
Figure 5.8 Import from LDAP screen
In this section are displayed configured LDAP remote servers, needed to download
phonebooks; such phonebooks will be available to users for browsing, research or click-tocall.
For each server user can: see last update time or import it again by clicking on the dedicated
button.
Moreover, user can modify settings for each LDAP server by clicking on it.
Figure 5.9 LDAP server settings
In the ‘LDAP Server Settings’ section user can define the following parameters:

Name
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
Host/Uri: LDAP server hostname or IP address

Port: if empty PBX will use the default port

RDN: the Root DN of the LDAP tree

LDAP version

Login credentials (if required): if empty, login will be anonymous.
From the LDAP server, PBX imports all the entries related to the research (objectClass =
person), extracting the following attributes:

givenName

surname

telephoneNumber

o

ou

email.
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Chapter 6 -
CDR
6.1 Call Detail Record
Figure 6.1 Call Detail Record
In this tab, the administrator can see the CDRs related to incoming and outbound calls
managed by KalliopePBX. Each month, a new tab is automatically created in order to simplify
the consultation. The ‘Current’ tab shows current month calls only. In order to display
previous months CDRs, user can click on the relative tab (for the previous six months) or find
it using the drop-down menu.
By default, PBX lists calls in reverse date order (newest to oldest) and reports details about:
o
uniqueID, call identifier
o
call type (incoming, outbound or local)
o
status (answered, not answered, busy, not available or forwarded to the voicemail)
o
Date and hour of arrive, answer (if answered) and hangup
o
caller ID (for incoming calls, if available),
o
called user ID (in case of incoming calls addressed to a call group, both the group and
the extension that actually answered are specified),
o
used gateway (for outbound calls)
o
Accountcode (if configured), available only with the optional Call Center module
o
call duration.
By clicking on the column header, user can reverse the order with respect to the selected
parameter.
In addition, user can filter CDRs for each of these fields; for example it is possible to restrict
the displayed data to: calls from a particular extension, or to a specific number, or in a given
period of time (set by clicking on the calendar icon in the Date/Hour column). Such selections
can be combined to create complex filters: after defining the filtering conditions, the results
will displayed by clicking on the ‘Filter’ button. To remove the filter just click on the ‘Clear’
button.
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Accessing at admin level, external phone numbers will be masked. It is possible to display
them entirely by entering the privacy password and selecting the checkbox ‘Unhide complete
numbers’ in the ‘Privacy Mode’ frame. Thanks to the privacy password it is also possible to
delete old CDR stored on the PBX by clicking on the dedicated button that appears once the
password is provided.
Accessing at user level, CDR tab will only display phone calls related to such user (without
any masking). In addition, next to external numbers, user will find several buttons that allow
him to: click-to-call, add contact to personal phonebook, add contact to shared phonebook.
Figure 6.2 Personal CDR
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Chapter 7 -
Call Recording
User can record phone calls (both incoming or outbound) in two ways:

Unconditional: the entire call is recorded

On demand: recording can be started or stopped by typing *1.
A specific audio message can be played when recording starts or stops.
Figure 7.1 Call Recording  Call Recording settings
From the ‘Call Recording Settings’ section user can enable/disable unconditional and/or on
demand recording and enable/disable call recording for internal phone calls. Enabling call
recording from this tab does not automatically imply its availability for all the extensions;
indeed, it is necessary to enable call recording individually for each extension, group, or ACD.
In case of extensions, call recording is available for both, incoming and outbound calls; on the
contrary, in case of call groups or ACD, call recording is available only for incoming calls.
The ‘Announcements’ frame allows the user to configure audio messages to play in order to
inform that call recording is activated or stopped. During the audio message, the extension
will hear a hold-on music.
Recorded calls can be stored both, on KalliopePBX storage or on an external memory
connected to the KalliopePBX USB port. KalliopePBX supports a single removable storage at
once (USB pen drive or external HDD), containing a single partition formatted as FAT32. The
‘Devices’ sections provide information about available storage devices (internal or external)
and relative free space.
Recorded calls can be downloaded, deleted or transferred from internal to external memory,
by clicking on the tabs ‘Local Storage’ and ‘USB Storage’. Such actions can be selected from a
drop-down menu and applied to the selected recordings; in order to perform the desired
action just click on the ‘Go’ button.
Figure 7.2 Local storage screen
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Chapter 8 -
Troubleshooting
‘Troubleshooting’ tab provides the user with several useful tools. One of such tools is a pcaplibrary-based tool that permits to capture traces of incoming/outgoing packets whitin
KalliopePBX network interface; optionally user can decide to apply a filter based on:
•
Protocol type (all or a combination of ICMP/UDP/TCP)
•
IP address (source or addressee)

Port (source or addressee).
Figure 8.1 Troubleshooting screen
When the filter is set, just click on the ‘Start’ button; capture can be stopped, manually, by
clicking on the ‘Stop’ button, or automatically, when captured file’s size reaches 10 MB.
Figure 8.2 Processing capture
Once the capture is concluded, user can download the pcap file by clicking on the specific
button. Pcap file is stored on KalliopePBX until: it is explicitly deleted by clicking on the
‘Delete’ button, a new file is captured or PBX is rebooted.
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Chapter 9 -
Provisioning
KalliopePBX is provided with a provisioning engine able to create (and send to phones thanks
to the built-in TFTP server) configuration files containing all the parameters needed to make
them function properly (PBX address, authentication credentials, etc…). At the moment,
KalliopePBX supports the phone models shown in the table here below.
Avaya
Gigaset Pro
Linksys
Yealink17
Snom18
T19
1220
DE310/410
SPA921/2
1230
DE700/900
SPA941/2
300/320/370
T20/21/22/26/28
710/715/720/760
T32/38
820/821/870
T41/42/46
VP530
Table 9.1 Provisioning, Supported phone models
Provisioning process involves three main steps:
1. Definition of the extensions to associate with phones (as described ins the panel PBX 
Extensions)
2. Device list (devices are identified by their MAC addresses)
3. Loading of one or more configuration templates, each for a specific phone model,
preconfigured with options common to various users classes. KalliopePBX can manage
more templates for each phone model in order to use different configurations for
different users classes using the same phone model.
When this process is finalized, KalliopePBX creates a configuration file for each phone model
associating the device’s MAC address to a specific extension and template. Such configuration
file is stored into the root folder of the built-in TFTP server; telephones can download it and
auto-configure themselves.
9.1 Auto-provisioning procedure
Once the configuration file is available on the TFTP server, the self-provisioning procedure is
as simple as connecting the phone to the network and turning it on.
Usually19, at first boot (or after a reset to factory settings), VoIP phones search for a DHCP
server to download their network configuration file. DHCP server must be able to send them
the configuration parameters (IP address, subnet mask, gateway address, etc.) and, in
addition, the Option 66, which is the TFTP server IP address needed by the phones to
download their configuration file.
When network configuration is operative, VoIP phones send a ‘Get’ request to the TFTP
server (set on the DHCP server equal to KalliopePBX IP address) in order to download their
configuration file. Such configuration is a text file whose name depends on the model of the
17
Tested with firmware release V50 and following.
Tested with firmware release 8.x.
19
For further details please see user manual for each specific phone.
18
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phone, but, contains the specific device MAC address. In case of Yealink phones,
configuration files are named 1234567890ab.cfg, while for Linksys phones files are named
spa1234567890ab.cfg; in both cases, 12:34:56:78:90:ab represents the MAC address. Snom
phones require, previously, a XML file named snomxxx.html (where xxx is the phone model,
for example 300, 320, 370), and, then, a file named snomxxx-12345678d90AB.htm.
Once the configuration file has been successfully downloaded, phones apply new
configuration, reboot and, eventually, are ready for use.
9.2 Provisioning procedure steps
9.2.1 Extensions creation
Please see paragraph 3.1.
9.2.2 Devices configuration
In Provisioning  Devices, click on the link ‘Add new device’.
Figure 9.1 Provisioning procedure, devices configuration
User must define the following fields:
•
MAC Address: device MAC address in one of these formats 1234567890ab,
12:34:56:78:90: ab, 12-34-56-78-90-ab
•
Host: whether the phone must have a dynamic or static IP address (in this case, it is
necessary to provide IP address, mask and Gateway)
•
DNS: optional, DNS server IP address that will be used by the phone
•
Model: phone model, to select from the drop down menu.
Figure 9.2 Adding new provisioning device
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9.2.3 Configuration Template definition
In Provisioning  Templates, click on the link ‘Add new templates’.
Figure 9.3 Adding new template
User must define the following fields:
•
Name: template name
•
Model: phone model, to select from the drop down menu
•
File: upload of a configuration template compatible with that specific model (such
file must have a .tpl extension). Template format changes depending on the phone
model. Please see the following paragraph for further details.
Figure 9.4 Loading a configuration template
When a configuration template file is loaded, it automatically appears in the list of
templates.
Figure 9.5 Template management screen
By clicking on the corresponding buttons, use can download, duplicate, edit or remove a
template.
Here below a template editing screen:
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Figure 9.6 Built-in editor for online template editing
9.2.4 Configuration file creation
In the tab Provisioning  Provisioning user can assign a specific device (identified by MAC
address) to an extension and a template by clicking on the link ‘Add new entry’.
Figure 9.7 Device/Extension/Template connection
‘Add new provisioning connection’ page is displayed and user must select from drop-down
menus:
•
Device (only those with no connections are displayed)
•
Extension (among those defined on KalliopePBX)
•
Configuration template (please pay attention, select the template associated to the
specific device model).
By clicking on the ‘Save’ button, the connection is created; KalliopePBX provisioning engine
creates the configuration file and stores it into the root folder of the built-in TFTP server.
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Figure 9.8 Provisioning configuration files management screen
Current connections are listed in the ‘Provisioning’ frame; user can select them and perform
following actions:
•
Temporary disable (the connection remains, but the configuration file is removed
from the TFTP server)
•
Re-enable (whether it was previously disabled)
•
Delete (the association is completely deleted, the device can be now associated to a
different extension)
•
Create (the configuration file is created again, useful if some account parameters
have been changed or the template is modified)
•
Connect the device to a template.
The entire list of associations is filterable or sortable using the search fields or by clicking on
the headers of the columns highlighted in orange.
9.2.4.1
Provisioning list import
A provisioning list can be imported from a XLS/CSV file containing the desired connections.
Such file must contain the following columns (header included):
•
"MAC": device MAC address
•
"EXTEN": extension associated to that specific device
•
"IP": device IP address (if empty or 0.0.0.0, DHCP will be used)
•
"NETMASK": Netmask Address
•
"GATEWAY": Gateway Address
•
"DNS": DNS Server Address
•
"DEVICE_MODEL": device model ID, according to the list provided in the import page.
Once the import process is complete, user must associate VoIP devices to the relative
template in order to create configuration files.
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9.2.5 Yealink phones templates
For Yealink phones, templates contain special codes (placeholder) replaced by the
provisioning system with the actual values. The currently supported placeholder are listed in
the following table:
Key
Placeholder
Value
Label
%YLKEY_LABEL%
Account label (First Name + Last Name)
DisplayName
%YLKEY_DISPLAYNAME%
Display Name (First Name + Last Name)
AuthName
%YLKEY_AUTHNAME%
SIP Account Auth. Name (Extension #)
UserName
%YLKEY_USERNAME%
Password
%YLKEY_PASSWORD%
SIP Account Password (Exten password)
SIPServerHost
%YLKEY_SIPSERVERHOST%
KalliopePBX Address
SIPServerPort
%YLKEY_SIPSERVERPORT%
KalliopePBX SIP Port (5060)
OutboundHost
%YLKEY_OUTBOUNDHOST%
KalliopePBX Address
OutboundPort
%YLKEY_OUTBOUNDPORT%
KalliopePBX SIP Port (5060)
WANType
%YLKEY_WANTYPE%
0: DHCP; 2:STATIC
WANStaticIP
%YLKEY_WANSTATICIP%
Phone IP Address (if WANType=2)
;[ account ]
;[ WAN ]
WANSubnetMask
%YLKEY_WANSUBNETMASK%
Subnet Mask (if WANType=2)
WANDefaultGateway
%YLKEY_WANDEFAULTGATEWAY%
Default Gateway (if WANType=2)
%YLKEY_DNS1%
Primary DNS (configurable per device)
;[DNS]
PrimaryDNS
Table 9.2 Placeholder available for Yealink provisioning templates
9.2.6 Linksys phones templates
For Linksys phone, configuration originates from a XML file 20 . Relevant tags, set by the
provisioning system, are (in case of multi-account phones, the first available account is
provisioned):
TAG XML
Value
User_Password
SIP Account ID (Extension)
User_ID_1_
SIP Account ID (Extension)
Password_1_
SIP account Password
DHCP
"Yes"/"No"
Static_IP
Phone IP address (if DHCP=”No”)
NetMask
Subnet Mask (if DHCP=”No”)
Gateway
Default Gateway (if DHCP=”No”)
Primary_DNS
Primary DNS
Proxy_1_
KalliopePBX IP address
20
Configuration file (to use as template) can be downloaded from Linksis phones web page at
the following address: http://<phone_IP_address>/admin/spacfg.xml.
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SIP_Port_1
pbxport
Outbound_Proxy_1_
KalliopePBX IP address
Display_Name_1_
Name Lastname;
Table 9.3 XML TAGs for Linksys phones provisioning
9.2.7 Snom phone templates
In case of Snom phones, user must provision the differences from the standard configuration
file. KalliopePBX provisioning system only generates snomxxx-1234567890AB.htm file for that
specific device.
Snom templates must use the following placeholders (that will be replaced during the
creation of configuration files):
TAG
Placeholder
Value
user_realname
%KP_DISPLAYNAME%
user_pname
%KP_AUTHNAME%
user_name
%KP_USERNAME%
user_pass
%KP_PASSWORD%
user_host
%KP_SIPSERVERHOST%
KalliopePBX Address
user_outbound
A%KP_OUTBOUNDHOST%
KalliopePBX Address
user_outbound
%YLKEY_OUTBOUNDPORT%
Outbound proxy Port (def. 5060)
Dhcp
%KP_WANTYPE%
on / off
ip_adr
%KP_WANSTATICIP%
Phone IP Address (if dhcp=on)
Netmask
%KP_WANSUBNETMASK%
Subnet Mask (if dhcp=on)
Gateway
%KP_WANDEFAULTGATEWAY%
Default Gateway (if dhcp=on)
%KP_DNS1%
Primary DNS (configurable per device)
mac address
%KP_MAC%
Device MAC address
mac address
%KP_MACADDRESS%
Device MAC address
;[ account ]
;[ WAN ]
;[DNS]
dns_server1
;[DEVICE]
Table 9.4 Snom provisioning templates placeholders
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Chapter 10 -
File Manager
KalliopePBX is provided with a built-in file browser that allows the user to upload files
straight into the TFTP server folder. In addition, KalliopePBX converts uploaded audio files
from wav/mp3 to G.711/G.729.
10.1 Audio Files
Figure 10.1 PBX  File Manager  Sounds
In this tab it is possible to manage audio files played by PBX to users (for example, busy
extension, IVR menu and submenu etc…).
Each section (IVR, opening time check, Miscellaneous) includes custom and default audio files
(not editable or erasable). Custom files can be uploaded, edited, renamed, moved or
deleted. PBX admits the upload of wav/mp3 files (PBX automatically converts them to a
telephone suitable format, such as PCM/mono/KHz). During the upload user can adjust the
volume by selecting from a drop-down menu a gain of 0, -3, -7, 0 -10 dB. Best quality can be
obtained by uploading audio files already sampled at 8 KHz.
Whether at least one G.729 license has been activated on KalliopePBX, it is possible to
convert audio files to such format too.
Uploaded audio files can be downloaded (in the new format) or played to a specific extension
in order to hear them straight from a device registered on the PBX.
Moreover, it is possible to record an audio message from an extension phone: user has to
specify the extension number and PBX will place a call towards such extension. User will be
asked to record a message and conclude the operation by pressing #. Once the message has
been recorded, user can play back the message and decide to save it or to record it again.
Finally, user can manage hold-on messages classes used in call ACD. For each class, the
configured audio files are played randomly or sequentially. One of such classes must be
tagged as the default one; it will be played in case of call transfer, call parking or on hold
calls.
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Rev.: 15-02-2015
On-call services codes
Such codes function only if typed by a PBX extension.
Code
Service
Attended Transfer: During a call between A and B, A types *4. PBX
plays a ‘transfer’ message and asks A to type C extension number
followed by #. In the meanwhile B is on-hold. Once A is connected
*4
to C, transfer can be finalized simply by hanging up and B is
connected to C.
Whether C does not accept the transfer, A can type *0 to restore
the initial call between A and B.
*0
Transfer cancellation.
Blind Transfer: During a call between A and B, A types #4. PBX
plays a ‘transfer’ message and asks A to type C extension number
followed by #. In the meanwhile B is on-hold. C’s device rings
(caller ID is B). When C answers he is connected directly to B.
#4
Whether C is busy or not available, the call will return to the
transferer and the number on the display will be preceeded by an
‘R’. A can restore the initial call with B only if C does not answer,
by typing *9 or **B.
*1
Start and stop on demand call recording.
Call Parking: During a call between A and B, A types #8. B is put
on-hold, an audio message will notify A in which slot the call has
#8
been parked (range P90 – P99). A can type the slot number (from
any extension) to recover the call.
After 45 seconds B has been parked, the call is forwarded back to
A.
**
Fast Transfer.
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Rev.: 15-02-2015
Appendix A: Provisioning
Yealink phones sample configuration template
#
#
#
#
#
#
#
#
#
#
#
#
#
Sample configuration template for Yealink T28P
Provisiong template for KalliopePBX
Supported versions:
v3.6.4
v3.6.5
Notes:
- 1 SIP account (all lines on the same account)
- NO VLAN
- SIP Registration Timeout : 5 minutes
- Tone set: Italy
- Timezone: IT - DST per week
[ account ]
path = /config/voip/sipAccount0.cfg
Enable = 1
Label = %YLKEY_LABEL%
DisplayName = %YLKEY_DISPLAYNAME%
AuthName = %YLKEY_AUTHNAME%
UserName = %YLKEY_USERNAME%
password = %YLKEY_PASSWORD%
SIPServerHost = %YLKEY_SIPSERVERHOST%
SIPServerPort = %YLKEY_SIPSERVERPORT%
UseOutboundProxy = 1
OutboundHost = %YLKEY_OUTBOUNDHOST%
OutboundPort = %YLKEY_OUTBOUNDPORT%
Transport = 0
BakOutboundHost =
BakOutboundPort = 5060
proxy-require =
AnonymousCall = 0
RejectAnonymousCall = 0
Expire = 300
SIPListenPort = 5060
Enable 100Rel = 0
MissedCallLog = 1
precondition = 0
SubsribeRegister = 0
SubsribeMWI = 0
CIDSource = 0
EnableSessionTimer = 0
SessionExpires =
SessionRefresher = 0
EnableUserEqualPhone = 0
srtp_encryption = 0
ptime = 0
ShareLine =
dialoginfo_callpickup =







Account label (First Name + Last Name)
KalliopePBX Address
KallipePBX SIP Port
 KalliopePBX Address
 KallipePBX SIP Port
 SIP registration expiration: 5 minutes
[ DTMF ]
DTMFInbandTransfer = 1
InfoType = 0
DTMFPayload = 101
[ NAT ]
NATTraversal = 0
STUNServer =
STUNPort = 10000
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Rev.: 15-02-2015
EnableUDPUpdate = 1
UDPUpdateTime = 30
rport = 0
[ ADVANCED ]
default_t1 = 0.5
default_t2 = 4
default_t4 = 5
[blf]
SubscribePeriod = 1800
BLFList_URI =
[ audio0 ]
enable = 1
PayloadType = PCMU
priority = 0
[ audio1 ]
enable = 1
PayloadType = PCMA
priority = 1
[ audio2 ]
enable = 1
PayloadType = GSM
priority = 2
[ audio3 ]
enable =
PayloadType =
priority =
[ audio4 ]
enable =
PayloadType =
priority =
[ audio5 ]
enable =
PayloadType =
priority =
[ audio7 ]
enable =
PayloadType =
priority =
[ audio8 ]
enable =
PayloadType =
priority =
[ audio9 ]
enable =
PayloadType =
priority =
[ audio10 ]
enable =
PayloadType =
priority =
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Rev.: 15-02-2015
[ WAN ]
path = /config/Network/Network.cfg
#WANType:0:DHCP,1:PPPoE,2:StaticIP
WANType = %YLKEY_WANTYPE%
WANStaticIP = %YLKEY_WANSTATICIP%
WANSubnetMask = %YLKEY_WANSUBNETMASK%
WANDefaultGateway = %YLKEY_WANDEFAULTGATEWAY%
[ DNS ]
PrimaryDNS = %YLKEY_DNS1%
SecondaryDNS =




0 for DHCP, 2 for static
Phone IP Address (if WANType=2)
Subnet Mask (if WANType=2)
Default Gateway (if WANType=2)
 Primary DNS (configurable per device)
[ PPPoE ]
PPPoEUser =
PPPoEPWD =
[ LAN ]
#LANTYPE:0:Router, 1:Bridge
LANTYPE = 1
RouterIP = 10.0.0.1
LANSubnetMask = 255.255.255.0
EnableDHCP = 1
DHCPStartIP = 10.0.0.10
DHCPEndIP = 10.0.0.100
[ VLAN ]
ISVLAN = 0
VID =
USRPRIORITY = 7
PC_PORT_VLAN_ENABLE = 0
PC_PORT_VID = 0
PC_PORT_PRIORITY = 0
 Bridge mode between WAN and LAN ports
 NO VLAN
[ QOS ]
SIGNALTOS = 48
RTPTOS = 48
[ RTPPORT ]
MaxRTPPort = 11800
MinRTPPort = 11780
[ AdminPassword ]
path = /yealink/config/Setting/autop.cfg
password = CHANGE-ME
devices)
 Change phone password (same for all
[ SYSLOG ]
#specify the server for syslog storage
SyslogdIP =
[ Lang ]
path = /config/Setting/Setting.cfg
ActiveWebLanguage = Italian
WebLanguage = Italian
[ Time ]
TimeZone = +1
TimeZoneName = Italy(Rome)
TimeServer1 = ntp1.ien.it
TimeServer2 = it.pool.ntp.org
Interval = 1000
SummerTime = 1
DSTTimeType = 1
TimeZoneInstead = 8
StartTime = 3/5/7/2
 Language of phone display
 Language of web gui
 Time, timezone and DST configuration
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Rev.: 15-02-2015
EndTime = 10/5/7/3
TimeFormat = 1
DateFormat = 0
OffSetTime = 60
DHCPTime = 0
[ PhoneSetting ]
InterDigitTime = 4
FlashHookTimer = 1
Lock = 0
Ringtype = Ring1.wav
Contrast = 2
BackLight = 2
BacklightTime = 30
ProductName =
[ AlertInfo0 ]
Text =
Ringer =
[ AlertInfo1 ]
Text =
Ringer =
[ AlertInfo2 ]
Text =
Ringer =
[ AlertInfo3 ]
Text =
Ringer =
[ AlertInfo4 ]
Text =
Ringer =
[ AlertInfo5 ]
Text =
Ringer =
[ AlertInfo6 ]
Text =
Ringer =
[ AlertInfo7 ]
Text =
Ringer =
[ AlertInfo8 ]
Text =
Ringer =
[ AlertInfo9 ]
Text =
Ringer =
[ Forward ]
path = /config/Features/Forward.cfg
Type = 0
AlwaysForward =
BusyForward =
NoAnswerForward =
AfterRingTimes = 10
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Rev.: 15-02-2015
Active = 0
BusyNoAnswerForward =
BusyNoAfterRingTimes = 10
[ Features ]
path = /config/Features/Phone.cfg
Call_Waiting = 1
Hotlinenumber =
BusyToneDelay =
[ PoundSend ]
#Set # key or * key as send. #:1 and *:2
Enable = 1
[ AutoAnswer ]
Enable = 0
[ Emergency ]
Num =
[ memory1 ]
path = /config/vpPhone/vpPhone.ini
#Set Memory key1
type =
Line =
Value =
PickupValue =
DKtype =




. . .
 Restore the memory2 to memory 9 contexts
From memory1 to memory10 are associated
to DSS keys (e.g. for BLF)
Available only in T26P and T28P
Can be removed from T20P, T22P templates
[ memory10 ]
#Set Memory key10
type =
Line =
Value =
PickupValue =
DKtype =
[ memory11 ]
key”
#Set line key1
DKtype = 15
Line = 0
Value =
 memory11 to memory16 are associated to “line
. . .
 Restore the memory11 to memory 15 contexts
 T28P has all of them
 T22P and T26P have 3 (memory11 to memory13)
 T20P has only 2 (memory11 and memory12)
[ memory16 ]
#Set line key6
DKtype = 15
Line = 0
Value =
[ Profile ]
path = /config/vpm.cfg
VAD = 0
CNG = 1
ECHO = 1
SIDE_TONE = -3
[ Jitter ]
path = /config/vpm.cfg
Adaptive = 1
Min = 0
Max = 300
Nominal = 120
- 105 -
Rev.: 15-02-2015
[ Message ]
path = /config/Features/Message.cfg
#Set voicemail number for each account
VoiceNumber0 =
VoiceNumber1 =
VoiceNumber2 =
VoiceNumber3 =
VoiceNumber4 =
VoiceNumber5 =
[ Country ]
path = /config/voip/tone.ini
Country = Italy
 Change Tone set
[ Tone Param ]
dial =
ring =
busy =
congestion =
callwaiting =
dialrecall =
record =
info =
stutter =
message =
autoanswer =
[ Default ]
dial = 1
ring = 1
busy = 1
congestion = 1
callwaiting = 1
dialrecall = 1
record = 1
info = 1
stutter = 1
message = 1
autoanswer = 1
[ AreaCode ]
path = /config/DialRule/areacode.cfg
code =
minlen =
maxlen =
[ BlockOut ]
path = /config/DialRule/BlockOut.cfg
#Set Block Out number.
1 =
2 =
3 =
4 =
5 =
6 =
7 =
8 =
9 =
10 =
[ RemotePhoneBook0 ]
URL =
Name =
URL =
Name =
- 106 -
Rev.: 15-02-2015
URL =
Name =
URL =
Name =
URL =
Name =
[ Webserver Type ]
path = /config/Advanced/Advanced.cfg
WebType =
Snom phones sample configuration template
<settings>
<phone-settings e="2">
<dhcp perm="">%KP_WANTYPE%</dhcp>
<ip_adr perm="RW">%KP_WANSTATICIP%</ip_adr>
<netmask perm="RW">%KP_WANSUBNETMASK%</netmask>
<gateway perm="RW">%KP_WANDEFAULTGATEWAY%</gateway>
<dns_server1 perm="RW">%KP_DNS1%</dns_server1>
<user_realname idx="1" perm="">%KP_DISPLAYNAME%</user_realname>
<user_name idx="1" perm="">%KP_USERNAME%</user_name>
<user_host idx="1" perm="">%KP_SIPSERVERHOST%</user_host>
<user_pname idx="1" perm="">%KP_USERNAME%</user_pname>
<user_pass idx="1" perm="">%KP_PASSWORD%</user_pass>
<user_outbound idx="1" perm="">%KP_OUTBOUNDHOST%:%KP_OUTBOUNDPORT%</user_outbound>
</phone-settings>
</settings>
Note: in case of DHCP device configuration, information related to static configuration (tags
ip_adr, netmask, gateway and dns_server1) is automatically removed from the configuration
file created by KalliopePBX Provisioning system.
- 107 -
Rev.: 15-02-2015

Administration Manual

Transcription

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