
UnlimCall Implementation Guides
Everything you need to connect your dialer, migrate off per-minute billing, and go live in any of 33 markets — in one place.
From $5/agent/day ($99/seat/mo, US/CA)
Dialer & PBX Setup Guides
UnlimCall connects to any SIP-capable dialer or PBX over a standard SIP trunk. No proprietary hardware, no locked firmware. Your agents keep the tools they already know; you replace the per-minute billing with a flat seat rate.
VICIdial
VICIdial is the most widely deployed open-source predictive dialer. Connecting UnlimCall takes roughly 20 minutes: register an outbound SIP trunk, set codec preferences to G.711u or G.711a, and point your carrier entry at UnlimCall's nearest edge POP. Caller ID is provisioned to your seat at onboarding — no additional config step.
FreePBX / Asterisk
FreePBX users add UnlimCall as a PJSIP or chan_sip trunk. The guide walks through trunk creation, outbound route priority, and codec order (G.711u first for US/CA, OPUS for latency-sensitive markets). STIR/SHAKEN attestation for US/CA numbers is applied at the network layer — no Asterisk module required.
GoHighLevel
GoHighLevel's hosted dialer uses SIP credentials for outbound calling. UnlimCall provides a dedicated SIP user per seat — credentials are issued during onboarding for each agent position you activate. The guide covers where to paste host, username, and password inside GHL's phone settings, plus how to verify registration status.
Custom SIP / Bring Your Own PBX
Running Kamailio, Yeastar, 3CX, or a homegrown softswitch? The custom SIP guide covers trunk authentication (IP-based or digest), recommended codec negotiation, SIP OPTIONS keep-alive intervals, and how to handle re-INVITE for mid-call audio path optimization. UnlimCall's edge accepts both TCP and TLS SIP transport.
Migrating from Per-Minute Billing
Switching from a per-minute carrier to a flat seat rate is a straightforward billing model change, but there are a handful of operational considerations: porting existing caller IDs, adjusting dialer pacing to account for the cost structure shift, and reconciling invoices mid-month.
The migration guide covers:
- How to calculate your break-even seat count vs. current per-minute spend
- Steps for porting DIDs or provisioning new caller IDs in the same markets
- How to run parallel trunks during a phased cutover without double-billing
- What to expect from your first UnlimCall invoice
Caller ID by Country
UnlimCall provisions caller IDs on demand at onboarding across all 33 live markets. There is no pre-stocked pool — numbers are allocated to your account when you activate a market. Each country has its own regulatory and carrier requirements for outbound number presentation.
The country caller ID guide covers:
- Which markets support local-format DIDs (geographic numbers vs. national numbers)
- Regulatory requirements for number registration before use (Germany, France, Australia, and others require end-user documentation)
- How caller ID presentation interacts with CNAM databases in the US and Canada
- Local presence dialing mechanics — presenting a local area code to the called party
STIR/SHAKEN for US and Canada
STIR/SHAKEN is a call authentication framework mandated in the US and Canada. UnlimCall signs outbound calls on US and Canadian numbers with Full Attestation (A-level) where origination conditions are met, helping calls display as authenticated on compliant terminating networks.
The guide explains attestation levels, what "signed" means on a receiving carrier's screen, and how to configure your dialer to preserve SIP Identity headers through your PBX rather than stripping them.
How Guides Are Organized
Each integration guide follows the same structure:
- Prerequisites (SIP client version, network requirements)
- Credentials you receive at onboarding
- Step-by-step trunk configuration with screenshots or config file snippets
- Verification steps (confirm registration, place a test call, check audio quality)
- Troubleshooting (common SIP error codes and their fixes)
If you are connecting a dialer not listed above, start with the custom SIP guide — the trunk parameters are identical regardless of frontend.