HexaVox Voice Setup
Call Center Workspace Email Guide

Voice Setup Guide

Set up browser calling, SIP softphones, forwarding, voicemail, and call routing in HexaVox.

Use this guide when a user needs to answer calls in the browser, connect a softphone, forward calls to mobile, or validate the account is ready for live voice traffic.

Browser Voice SIP Softphone Go-Live Checklist
Use the HexaVox browser call center

Best choice for users who want to answer, dial, and monitor calls directly inside HexaVox.

 Connect a SIP softphone or desk phone

Use this when the team prefers Zoiper, Linphone, or another SIP-capable client.

 Use mobile forwarding and voicemail fallback

Useful when calls should ring a real phone after or instead of the browser softphone.

 Configure IVR, ring groups, and fallback routing

Use this for inbound call menus, team queues, after-hours handling, and operator routing.

Path 1

Use the HexaVox browser call center

Recommended for users who want the simplest answer-and-dial experience without installing a separate softphone app.

What the admin must enable first

  1. Assign an account voice number. The account needs at least one active caller ID.
  2. Enable voice access for the user. Turn on receiving and, if needed, outbound calling.
  3. Provision voice provider. HexaVox production voice routes through Telnyx. Confirm provider health and credentials are live.
  4. Choose the inbound route. Set a direct user, ring group, or IVR as the primary route.

What the end user does

  1. Open /calls. This is the live voice workspace.
  2. Allow microphone access. Use HTTPS and confirm the browser is not blocking the microphone.
  3. Click Connect Browser Voice. Wait until the status shows the softphone is ready.
  4. Run one inbound and one outbound test. Verify ring, audio, caller ID, and hangup behavior.

Best fit. Browser voice is the easiest setup for most office users because it keeps queue status, inbound answer, outbound click-to-call, and follow-up context in one place.

Provider note. Production voice routes through Telnyx. If the call center shows degraded status, treat it as an ops/provider issue rather than retrying calls repeatedly.

Path 2

Connect a SIP softphone or desk phone

Use this when users prefer a dedicated SIP client such as Zoiper, Linphone, a browser SIP app, or a SIP desk phone.

Admin steps in HexaVox

  1. Enable SIP access for the user. Generate the SIP password inside HexaVox settings when needed.
  2. Confirm the extension. The user should have a voice extension or SIP identity assigned.
  3. Choose UDP or TLS. TLS is preferred when the client supports it.
  4. Test registration before go-live. Make sure the client registers and receives a live inbound test.

Generic SIP settings

  • Host sip.gohexavox.com
  • Domain sip.gohexavox.com
  • UDP Port 5060
  • TLS Port 5061
  • WebSocket wss://sip.gohexavox.com:7443
  • Username HexaVox SIP username or assigned extension identity
  • Password HexaVox SIP password generated in the workspace

Security defaults. Prefer TLS 5061 when the client supports it. Use UDP only on trusted networks.

Use the right mode. If the user only needs the built-in HexaVox call center, browser voice is simpler. Use SIP when the team has a clear reason to stay on a dedicated softphone or desk phone.

Path 3

Set up mobile forwarding and voicemail fallback

Use this when calls should ring a real mobile phone, or when you want a backup route after the browser or SIP client does not answer.

Forwarding setup

  1. Add the user’s forward number. Use full E.164 format such as +15551234567.
  2. Choose when it rings. Decide whether the mobile should ring immediately, after the browser, or only after-hours.
  3. Test caller ID. Confirm the external leg still shows the expected HexaVox account number.
  4. Document fallback behavior. Make it clear whether no-answer goes to voicemail, another user, or another number.

Voicemail setup

  1. Turn on voicemail delivery. Choose email, SMS, WhatsApp, or another configured notification path.
  2. Set the greeting. Use the account’s default greeting or upload a custom prompt.
  3. Confirm after-hours behavior. Route closed-hours calls to voicemail or a dedicated mobile fallback.
  4. Run a missed-call test. Confirm the voicemail arrives where the team expects it.

Path 4

Configure IVR, ring groups, and fallback routing

Use this when an account needs a real inbound call flow instead of just ringing a single user directly.

Ring Groups

  • Create a named group such as Support or Sales.
  • Pick the routing strategy: simultaneous, round robin, priority, or longest idle.
  • Set the timeout and choose whether missed calls should fall through to voicemail.

IVR Menus

  • Create the greeting and decide whether text-to-speech or uploaded audio should play.
  • Map keys like 1, 2, and 0 to groups, users, or direct numbers.
  • Set fallback, after-hours, and timeout behavior so calls do not get stranded.

Primary Route

  • Assign a primary IVR or direct route for the account’s main number.
  • Run one inbound business-hours test and one after-hours test.
  • Document the live call flow before users go into production.

Keep the routing simple at launch. One main line, a short greeting, two or three menu choices, and a clean fallback path is usually better than a deep menu tree.

Go-Live

The voice readiness checklist to use before production traffic

Account readiness

  • The account has at least one active voice number.
  • The right users have receive and outbound voice permissions.
  • Primary routing is assigned and documented.
  • Caller ID is verified on a real outbound test.

Operational readiness

  • Inbound answer works in the browser, SIP client, or forwarding target.
  • Voicemail fallback works when calls are missed.
  • After-hours behavior matches the business rule.
  • The team has run one inbound and one outbound live test on production configuration.

Reconcile-first safety. If a provider acceptance is ambiguous, HexaVox blocks blind retry until reconciliation completes. If you see delayed callbacks, wait for the status to reconcile instead of re-dialing manually.

Terminal state is immutable. Once a call reaches COMPLETED, FAILED, or VOICEMAIL, HexaVox ignores late non-terminal provider noise instead of reopening the interaction.

Troubleshooting

The first places to check when voice setup fails

Browser will not connect

  • Confirm microphone permission is allowed.
  • Stay on HTTPS and refresh the call center.
  • Reconnect browser voice after any token or network interruption.

SIP client will not register

  • Check the SIP username and the current one-time password.
  • Try TLS first if the client supports it.
  • Confirm the client is pointing at the HexaVox SIP host and not an old provider hostname.

Calls are routing incorrectly

  • Review the primary IVR or direct route.
  • Check ring group membership and agent status.
  • Verify after-hours, fallback, and voicemail rules with live tests.