Professional PBX Solution

Main features

  • Full compatibility with all  SIP standard devices 
  • Up to 10 concurrent calls and 12 users / extensions configurable
  • Up to 4 telephone lines
  • Firmware updates completely free
  • No cost to be Renewed in time
  • Complete with all the capabilities of modern PBX
  • FAX support (T38, passing)
  • Personal Address Book LDAP and web-based system with click-to-dial
  • Transport SIP over TCP and TLS. Encryption via SRTP streams.
  • WebAPI and Rest APIs for integration with third party applications.
  • Call recording (unconditionally and on request)
  • Provisioning of SIP telephone equipment (Cisco, SMB, Gigaset Pro, Snom and Yealink)
  • Support for SIP video (H263 / H264)
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Product Description

SICEPBX by  KalliopePBX renews the concept of telephony, both Compared to the traditional telephony to IP telephony, offering a range of VoIP system combines advanced technical features with easy to use and manage

SICEPBX-MINI by  KalliopePBX  is the entry-level VoIP PBX Specifically designed for micro-enterprises and For All Those Who need 4 lines with up to 12 extensions / users.

SICEPBX-MINI by  KalliopePBX  Provides up to 10 simultaneous calls and thanks to the use of solid-state storage technology fanless and can be Placed in a closed, dust or vibrations without any danger for its operation.

SICEPBX-MINI by  KalliopePBX  Allows, in case of increase of customer needs, to move to SICEPBX-LITE by  KalliopePBX  simply changing the license on board apparatus. This Allows a considerable upgrade the VoIP network (20 concurrent calls and unlimited users) without hardware changes .

SICEPBX by  KalliopePBX  has a web interface configuration designed to be used by Those without a specific VoIP expertise. And thanks to the simple (and free) firmware upgrade the PBX will always be updated. The main feature of SICEPBX by  KalliopePBX are Transferring, forwarding and call back, support for video calls, provisioning, configuration, service availability, Call Parking, Transport SIP over TCP and SSL, personal address book and system Web Based, Web API integration with third-party applications, Call Record and ACL, Padlock electronic and voice mail address by sending voice messages via email.


System Integrator


Network Operator



Max User 12
Phone lines 4
Contemporary Calls 10
Rackmount No
Call Tranfer With or without offer
Call Forwarding Unconditioned
Call back On busy/ no answer
Call park Yes
Voice responder Yes, interactive (IVR) Multilevel
Day/night service Yes, multiple
Call Detail Record Yes, with filters and CSV export
Voice mail Yes, voice mail by sending voice messages by e-mail
Audioconference Yes, multi-user / multi-room
Availability Service Yes, for system, for group and for each user
Withdrawal Call Yes, explicit and for groups di prelievo by invitation or direct
DISA Service Yes
Callback Service Yes
Outgoing lines routing Customer policy based
Electronic lock Yes
Fixed-mobile integration Yes, with service dual-ring combined with the Fast-Transfer
Voice recorder Yes, Unconditioned and by request, in input and output
Advanced management of queues Yes
ACL Yes, access control on IP of extensions
TXT messages management Yes, with external ip provider or with GSM gateway
BLF support Yes (Busy Lamp Field)
Videocalls Yes, SIP (H263/H264)
Provisioning SIP Yes, for Cisco SMB, Gigaset Pro, Snom and Yealink
Address book Yes, personal and centralized web based system with click-to-dial and LDAP synchronization
Extention booking Yes, on Busy or without answer
Transport TCP and TLS, ciphering via SRTP
FAX Support Yes, T38
Third Party Integration Yes, via Web API e API Rest

Ordering information


VoIP Unit max 12 users and 4 lines.


Uograde License from MINI to LITE