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Key features

  • 2 SIP accounts
  • Autoprovision (TR-069, DHCP)
  • Headset connection
  • Large LCD display

VP-12 – up-to-date IP phones with integrated switch, which provides PC and IP phone connection through one physical line.

Business solution
VP-12 are designed for companies with high quality voice data transmission requirements. The IP phones also provide reliability of connections and usability.

High quality of sound
The high performance of VP-12 phones is provided by the hardware based on advanced Realtek chip. All the main audio codecs used in VoIP networks (G.711, G.722, G.723.1, G.726, G.729) are supported by the VP-12 phones. Echo cancellation, Silence Detector, DTMF signals reception and generation as well as traffic prioritization (QoS) ensure high quality of voice data.

Eltex.ACS management system (TR-069)
Auto Configuration system Eltex.ACS based on TR-069 specification provides easy-to-use functionality for firm- ware versions management and phone configuration. Eltex.ACS loads individual configuration to the device automatically according to the subscriber credentials. Intellectual firmware updating function works consistently with devices on the network, updating device groups according to the settled queue. Eltex.ACS has NorthBound Interface for OSS integration.

Convenient key location, intuitive menu and user-friendly management web interface with multilingual support provides usability in corporate telephone networks.


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  • Phone functions

    • 2 SIP accounts configuring independently
    • Geographical redundancy of a SIP server (support for up to 4 SIP redundant servers)
    • Flexible dialplan
    • Operation without a SIP server
    • Caller name and number displaying (Caller ID)
    • Mute
    • Redial
    • Different ring-tones for different accounts
    • Possibility to upload ring-tones to the phone
    • Call history
    • Local phonebook for 200 phone numbers with export and import feature
    • LDAP Remote Phonebook
    • Speakerphone mode
    • Short text messages transmitting and receiving (SIP MESSAGE)
    • View voice mail counters
    • Indication on waiting voice messages (MWI)
    • Remote phonebook

    Value Added services

    • Call Hold
    • Call Transfer
    • Call Waiting
    • Call Forward on Busy
    • Call Forward on no response
    • Call Forward unconditional
    • Do not Disturb mode (DND)
    • Caller Line Identification Restriction (CLIR)
    • Local 3 Way-conference
    • Support for remote conference according to RFC4579
    • Hotline/warmline
    • Intercom calls acceptance

    Display and indicators

    • 3.2 inch backlit monochrome display, 128x64 resolution
    • Interface language selection (English or Russian)
    • Functional keys with LED indication: accounts’ keys, message, do not disturb, mute
    • Functional keys without LED indication: call transfer, call hold, conference, redial, headset, speakerphone, volume adjustment
    • 2 SIP lines keys with status indicators

    VoIP protocol

    • SIP

    Audio codec support

    • G.711 a-law, µ-law
    • G.723.1
    • G.726
    • G.729

    Voice standards

    • Voice activity detection (VAD)
    • Echo Cancellation, G.165, G.168 (AEC)


    • Device status monitoring via the web interface
    • Displaying of debugging information in Syslog, Telnet

    Monitoring and management

    • Management web interface (Russian and English versions)
    • SSH
    • Telnet
    • Autoprovision (TR-069, DHCP)

    Network protocols and security

    • Data network connection: Static, DHCP, PPPoE, No IP
    • Time and date synchronization via NTP
    • 802.1p, DSCP traffic marking
    • Support for 802.1X
    • Support for NAT traversal: STUN mode, Public IP
    • Support for LLDP, LLDP MED
    • Support for SIP over TLS
    • Support for SRTP
    • LDAP over TLS


    • RFC 3261 SIP 2.0
    • RFC 3262 SIP PRACK
    • RFC 4566 Session Description Protocol (SDP)
    • RFC 3263 Locating SIP servers for DNS lookup SRV and A records
    • RFC 3264 SDP Offer/Answer Model
    • RFC 3311 SIP Update
    • RFC 3515 SIP REFER
    • RFC 3891 SIP Replaces Header
    • RFC 3892 SIP Referred-By Mechanism
    • RFC 4028 SIP Session Timer
    • RFC 2976 SIP INFO Method
    • RFC 2833 RTP Payload for DTMF Digits, Flash event
    • RFC 3108 Attributes ecan and silenceSupp in SDP
    • RFC 4579 SIP Call Control - Conferencing for User Agents
    • RFC 3361 DHCP Option 120
    • RFC 3550 RTP A Transport Protocol for Real-Time Applications
    • RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)
    • RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
    • RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initial Protocol (SIP)

    Technical features

    • Realtek 8972C processor
    • SDRAM 128 MB
    • SPI Flash 16 MB
    • Linux software
    • Software redundancy

    Physical features

    • 2хRJ45 10/100Mbps Ethernet ports
    • 1хRJ9 (4P4C) for handset connection
    • 1хJack 3.5 mm or 1xRJ49 (4P4C) for hands-free connection1
    • Power consumption: no more than 3.5 W (maximum current consumption 0.7 A)
    • Power adapter: 5 VDC, 2A
    • Operating temperature: from +5°С to +40°С
    • Operating relative humidity at +25°С: up to 80%
    • Desktop case
    • Dimensions: 223x178x89,5 mm
    • Weight: no more than 0,52 kg
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