Service

Business VoIP Solutions

FreePBX, Asterisk, 3CX, and SIP trunking — number porting, IVR, queues, softphones and desk phones, QoS, and staged cutover with rollback.

  • FreePBX
  • Asterisk
  • 3CX
  • SIP
  • Kamailio
  • Yealink
  • Poly

Why Varnox

What you can expect

  • 01

    Platform chosen for your scale, self-host vs cloud preference, and licensing budget

  • 02

    Pilot extensions and trunk testing before full cutover; legacy lines kept until you are confident

  • 03

    Call-flow diagrams, extension directory, and admin guide for everyday changes

Full scope

Everything in this lane

High-level checklist of what we cover — scoped and priced per project, not all at once unless you ask.

Platforms & trunking

  • FreePBX, Asterisk, and 3CX — cloud-hosted or on-prem per your requirements
  • SIP trunk selection, registration, failover trunk, and outbound caller ID
  • Number porting coordination, LOA paperwork, and test DID before cutover
  • Kamailio or OpenSIPS for routing at scale when architecture requires it
  • International and toll-free number strategy where carriers support it

Call flows & features

  • IVR menus, time conditions, and holiday schedules
  • Ring groups, hunt groups, call queues, and agent login/logout
  • Voicemail, voicemail-to-email, and custom prompts (recorded or TTS)
  • Conference rooms, call parking, pickup groups, and intercom/paging where needed
  • After-hours routing to mobile, answering service, or voicemail

Endpoints & users

  • Desk phones (Yealink, Poly, etc.) provisioning and template configs
  • Softphones: 3CX app, Zoiper, Linphone, and mobile clients
  • Hot-desking, shared reception, and remote worker extensions
  • Headset compatibility and sidecar layouts for reception-heavy roles

Quality, security & compliance

  • Codec selection (G.711, Opus) and jitter buffer tuning
  • QoS/DSCP on business routers; bandwidth planning per site
  • NAT traversal, STUN, and firewall pinholes documented
  • SIP TLS/SRTP where provider and endpoints support it
  • Call recording setup with retention and consent notes for your jurisdiction

Integrations & handover

  • CRM click-to-dial or basic pop-up integrations when platforms allow
  • Microsoft Teams or legacy PSTN coexistence planning during migration
  • Admin training and written guide: add extension, change hours, update IVR
  • Monitoring trunk health and alerting before users notice dropped calls

Delivery

How we deliver it

Clear communication, documented changes, and pragmatic defaults — so you are never guessing what happens next.

  • Phone system changes fail when porting, firewall, and user training are afterthoughts. We stage pilots, test under real traffic, and keep rollback paths until you sign off.

  • You should manage everyday tasks — new extension, holiday hours — without opening a support ticket for each change.

FAQ

Common questions

About Business VoIP — ask us anything on the contact form if you do not see your scenario.

FreePBX, Asterisk, and 3CX primarily, plus Kamailio for SIP routing at scale. Choice depends on self-host vs cloud, licensing, and extension count.
Usually yes through your carrier or SIP trunk provider, with LOA coordination and testing before cutover.
Not necessarily — many teams start with softphones. Compatible desk phones are recommended when hardware makes sense, without overselling features.
Codec tuning, jitter buffers, and router QoS/DSCP are in scope. Home broadband limits are stated upfront; trunk monitoring helps catch provider-side issues early.
Yes — IVR, ring groups, call queues, voicemail-to-email, time conditions, and after-hours routing, all tested before full cutover.

Discuss this service

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