+420 211 151 657helpdesk@insoft.czflag of the United States


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Smart automatic operator

This system monitors the calls that have taken place and, based on the caller's identification, is able to automatically connect the call to the line that was last connected to the caller. Thanks to this function, it is possible to, for example, fully use GSM gateways for outgoing and incoming calls. It is usually necessary to block caller identification for outgoing calls so that the called party does not attempt to call back the GSM gateway number in the event of a missed call. Otherwise, the person handling incoming calls from GSM gateways would not know who the caller was trying to call and to whom to transfer the call. When using a smart automatic attendant, the call is automatically connected to the line that was last connected with the caller.

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Hierarchical numbering plan

We place great emphasis on enabling our clients to easily set up and operate their systems. Therefore, all telephones, voice answering machines, conferences, state lines and other system components are classified in a tree structure that faithfully replicates the company's organizational structure. Each group can have certain parameters assigned to it (e.g., range of line numbers, filter for outgoing calls, etc.), thanks to which adding a new phone means only selecting the group and filling in the user name and phone type. This minimizes the time required to set up a new employee's line and completely eliminates the cost of a service technician. All other components of the system work with the tree structure, so it is easy to set up access to call billing by departments or companies that share a single PBX.

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'Follow Me'

The Follow Me feature is especially suitable for WiFi IP phones and mobile phones with WiFi and SIP support. The system automatically detects the presence of a portable phone on the company premises. If the user is not logged in to his/her landline phone, incoming calls are routed to the user's mobile phone via the free WiFi network.

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Backup trunk notification

In the event of a main trunk failure, it is possible to set a voice message that will be played to the caller as a warning of the unavailability of the main trunk. The user can thus decide whether the call is important and make it through a backup trunk, which, for example, does not provide the same capacity or as favorable a tariff as the main trunk.

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User portal

Normal PBX users can access the individual components of the system via the web interface. This makes it easy for them to get an overview of past and missed calls and access voicemail messages and call accounting. Based on the permissions assigned by the system administrator, they can also set up call forwarding in case of absence or based on attendance status, including call forwarding to the public telephone network. The user portal is ready for integration with other company systems using server-side and remote plug-ins. Single-Sign-On functionality is supported for easy access.

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Supported protocols

  • signaling: SIP, H.323, IAX, MGCP, SCCP, Skype
  • voice codecs: ADPCM, G.711 (A-Law, u-law), G.722, G.723.1, G.726, G.729, GSM, iLBC, Linear, LPC-10, Speex, T.38
  • integrative: XML-RPC, SOAP, CSTA, AGI, SNMP
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Call billing

  • filtering and grouping the listing based on caller number, called party, line, group and time period
  • Tariff prices can be entered for each trunk, and the system then automatically calculates prices for individual calls.
  • restriction of billing access for individual phones and groups of phones based on user rights (grouping)
  • distinguishing business and private calls according to the corporate directory
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  • access to voicemail using a web browser, XML application in IP phone and IVR (including access from public telephone network). Notification of a new message can be received by (either or both):
    • e-mail with the option of delivering a recorded message in WAV or MP3 format and deleting the message after delivery
    • new message indication on your phone
  • the possibility of restricting access to the mailbox by PIN
  • unlimited number of mailboxes
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Contact center

  • waiting queues with priority support
    • limiting the maximum number of people waiting
    • operator can operate multiple queues
    • operator login to the queue or permanent service
    • adjustable opening announcement, music while waiting and announcements about the number of people waiting
  • distribution algorithms: linear, circular, longest non-speaking, broadcast
  • supervisor client application
    • statistics of queue occupancy and waiting times
    • operator workload overview
    • listening to calls
  • client application for operators
    • display of caller ID and waiting queue
    • control of operator functions
    • remote control of the phone from the application
    • data retrieval from external source/application
    • external application control
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Call recording

  • recording of external and internal calls
  • access to recorded calls included in call billing
  • permanent or on-demand recording
  • filtering rules to limit uploading
  • MP3 and WAV formats supported
  • archiving of calls, including call information
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Unified messaging

  • receiving voice messages as e-mail
  • API for receiving and sending SMS messages
  • sending SMS messages from GSM gateways to e-mail
  • Jabber protocol support
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Presence and mobility

  • line registration on any phone (extension mobility)
  • PIN security for outgoing calls
  • option to redirect the line to the public telephone network
  • calls via the system from the public telephone network (DISA)
  • user status (available, do not disturb, unavailable, out of office, not logged in)
  • application personal assistant
    • tree list of participants in the system (telephone lines)
    • setting and displaying the presence status
    • instant messaging (Jabber)
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  • hierarchical numbering plan with support for identical line numbers (partitioning)
  • outgoing call
    • direct support of SIP trunks including support for registration to the provider's server (SIP registrar)
    • trunk selection according to the cheapest price (LCR)
    • ENUM support
    • call blocking based on the called number
    • support for automatic switching to a backup trunk in the event of a main failure
    • swapping of caller and called numbers
    • temporary and permanent hiding of caller identification (CLIR)
    • limiting the maximum call length
  • incoming call
    • direct support of SIP trunks
    • routing based on calling number, called number and time
    • call blocking (blacklist) with rejection (non-existent, busy or recorded voice)
    • swapping of caller and called numbers
    • caller ID display (CLIP)
    • looking up the caller name in the company database based on the caller ID (CID)
    • playing voices before connecting the call (early-media)
    • limiting the maximum call length
  • Transferring
    • forward all
    • forward busy
    • forward no-answer
    • forward no-coverage
    • forward do-not-disturb
    • if out of the office (forward out-of-office)
    • distinction between external and internal calls
  • transfer with consultation, or blind transfer (without consultation)
  • call hold)
  • call waiting
  • hunt-groups
  • call park
  • call pickup
  • music on hold
  • do not disturb
  • call-back
    • activation option only for selected lines
    • voice information to the caller about the possibility of call-back
  • auto attendant
    • possibility of chaining offers
    • routing according to the calls made
  • ad-hoc conference
    • connection of 3 participants
  • conference rooms (meet-me conference)
    • connection of up to 15 participants
    • the possibility of restricting access to the conference by PIN
    • conference administrator
    • automatic creation of the conference at the selected time, including periodic repetition
  • provisioning
    • support for Cisco and Linksys phones with the possibility of extension to phones from other manufacturers
    • XML application server for phones with XML browser support
    • phone books (possibility of connecting to several sources at the same time)
  • administrator access via web interface
    • system settings
    • IP telephony settings
    • system status monitoring
  • user access via a web interface with localization support (standard English, Czech)
    • individual functions are available according to the set permissions
    • online call overview
    • access to voicemail
    • voicemail settings
    • redirection settings
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System properties

  • user interface localization
    • Czech, English
    • the possibility of localization into any language using the GNU gettext standard
  • redundancy
  • full backup of all services
  • unified management (replication without primary/secondary distinction)
  • VLAN support (number unlimited by software)
  • integrated firewall (iptables)
  • integrated DHCP server
  • integrated VPN server (OpenVPN)
  • system monitoring using SNMP including support for SNMP traps
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Hardware requirements

The following parameters are based on reference HW 2x Intel Xeon Quad Core CPU, 4GB RAM, RAID 10 SAS, 2x 1Gbit NIC
  • Number of registered phones: 1 000
  • Number of concurrent calls: 200
  • Number of simultaneous calls including recording: 100
  • Number of queues in the contact center: 25