Support / Voice / Phone

Hardware IP phone configuration.

Most SIP desk phones expose the same handful of account fields. The defaults differ vendor to vendor; the values to set them to don't.

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Common fields, recommended values

  • Transport — UDP (most common) or TCP.
  • SIP server / proxy — Shown in the customer portal under your trunk's settings.
  • Outbound proxy — Same value as the SIP server unless engineering specified otherwise.
  • Username / Auth name — Per-trunk SIP credentials (not the portal account).
  • Register expiry — 360 seconds.
  • NAT keepalive — Enable; 30 seconds. Most vendors call this "keepalive interval" or "NAT traversal."
  • DTMF — RFC 2833 (in-RTP). Use SIP INFO only if your application explicitly requires it.
  • Codecs — G.711µ (PCMU) primary, G.711a (PCMA) fallback, G.729.

Vendors

  • Yealink

    T46U, T48S, T54W, T57W, T33G — most common SOHO/business desk phones.

  • Polycom VVX

    VVX 250, 350, 450, 601 running UC Software 6.x.

  • Poly Edge

    Edge B10/B20/B30 and E series — Poly's post-VVX line running Poly OS.

  • Cisco MPP

    Cisco IP Phone 7800/8800 series running Multi-Platform (MPP) firmware.

  • Grandstream

    GRP series, HT80x ATAs, GXP series legacy phones.

  • Avaya

    J100 series (J139, J159, J169, J179, J189) running Open SIP firmware.

  • Mitel

    6900 series (6920, 6930, 6940, 6970) running SIP firmware.

  • Snom

    D7xx and D8xx desk phones plus M-series DECT bases.

  • Fanvil

    X-series desk phones, V-series video phones, i-series door phones, ATAs.

  • Sangoma

    S-series (S205–S715) and D-series (D60–D80) desk phones, plus FreePBX EndPoint Manager integration.