Sales Dialer Setup: Connecting Your CRM to a Flat-Rate SIP Trunk

The three layers that most teams conflate

Sales telephony infrastructure has three distinct layers that vendors routinely bundle together to create lock-in:

Layer 1 — The CRM. Where prospect data, call history, meeting records, and pipeline stages live. Salesforce, HubSpot, Pipedrive, Close, and dozens of others. This layer should be the source of truth for prospect data and activity logging.

Layer 2 — The dialer. The software that manages the outbound calling workflow — predictive dialing, power dialing, click-to-call, call queuing, voicemail drop, disposition logging. ViciDial, Five9, PhoneBurner, Kixie, GoHighLevel, and others occupy this layer.

Layer 3 — The SIP carrier. The network that actually completes the call. This is the layer that determines call quality, caller ID presentation, and carrier cost. Most teams accept whoever their dialer vendor has pre-integrated, often without realizing it is a separate commercial relationship.

These layers can — and often should — be separated. Choosing your dialer independently from your CRM, and your SIP carrier independently from your dialer, gives you the ability to optimize each layer without being constrained by another vendor's integration choices.

What a SIP trunk is and why it matters

A SIP trunk is the connection between your dialer platform and the public telephone network. When a rep clicks "dial" in their dialer interface, the dialer sends a SIP INVITE to the SIP carrier. The carrier completes the call to the public switched telephone network (PSTN) and passes audio in both directions.

The SIP carrier determines:

• Whether the call actually completes (reliability)
• The audio quality of the connection (codec support, latency, packet loss handling)
• What caller ID number is presented to the prospect
• The cost of the call (per-minute rate or per-seat flat rate)
• The geographic markets where calls complete at acceptable quality

When teams run their SIP carrier through their dialer vendor without examining it separately, they frequently accept all four of these variables as non-negotiable. They are not.

Connecting UnlimCall to common dialer platforms

UnlimCall provides SIP credentials (username, password, SIP server address) at onboarding. These credentials connect to any SIP-compatible dialer.

ViciDial. Add a new SIP carrier under Admin → Carriers. Enter the UnlimCall SIP server address, username, and password. Set the outbound caller ID to the local number provisioned for your account in the target market. ViciDial's predictive dialing engine then routes calls through the UnlimCall trunk.

FreePBX / Asterisk. Create a new trunk under Connectivity → Trunks → Add SIP (chan_pjsip) Trunk. Configure the registration string with UnlimCall credentials. Set outbound caller ID per dial plan. Outbound routes assign the trunk to the relevant extensions or ring groups.

GoHighLevel. GHL supports SIP outbound through its Phone Numbers section. Add a SIP trunk with the UnlimCall credentials. Assign local numbers to sub-accounts aligned to your geographic campaign targets.

Kixie / PhoneBurner. These platforms support custom SIP trunking in their advanced tier plans. Contact their support to configure an external SIP trunk using the UnlimCall credentials. Not all plans support external SIP — verify with your vendor before making commitments.

Custom Asterisk or FreeSWITCH builds. Provide the UnlimCall SIP server, credentials, and codec preferences (G.711 µ-law and G.729 supported) to your PBX administrator. The trunk configuration is standard SIP — no proprietary extensions required.

CRM integration: where call data flows after the SIP layer

The SIP carrier layer handles call completion and audio. The CRM integration is the dialer's responsibility — the SIP trunk does not need to know anything about your CRM.

Most dialer platforms log activity to CRM automatically via API: call disposition, call duration, recording link, notes from the rep. This integration should be configured at the dialer layer without modification when the SIP carrier changes.

When switching SIP carriers (including to UnlimCall), the CRM integration does not change. The dialer's behavior does not change. The only change is which network completes the call and what carrier cost appears on the monthly invoice.

This is the core benefit of carrier-layer separation: you can optimize carrier cost and quality without touching the rep workflow, the CRM integration, or the call recording setup.

Caller ID configuration in the dialer

Once local caller ID numbers are provisioned for your account in each target market, they need to be configured in the dialer to present correctly on outbound calls.

In most dialers, this is an outbound route or campaign setting: "present this caller ID for calls to this geographic target." For a US-focused team with local numbers in Chicago, Dallas, and New York, the dialer campaign is configured to present the appropriate area code for each geographic subset of the prospect list.

This configuration is dialer-specific. In ViciDial, it is set at the campaign level. In FreePBX, it is set in the dial plan via the outbound route caller ID override. In GoHighLevel, it is set per sub-account number. Your implementation specifics depend on your platform.

What does not change: the underlying principle. The number presented to the prospect is a local number provisioned to your account — not a number from a shared pool, not a number that rotates among tenants, and in the US and Canada, a STIR/SHAKEN attested number that carries a cryptographic origination signature.

Codecs, latency, and audio quality

Audio quality on outbound sales calls is a conversion variable, not just a technical footnote. A call where the prospect hears compression artifacts, echo, or delay takes longer to close, if it closes at all.

UnlimCall supports G.711 µ-law (PCMU) and G.711 a-law (PCMA) as primary codecs, with G.729 available for bandwidth-constrained environments. For outbound sales, G.711 is preferred — it is uncompressed and introduces minimal processing latency. G.729 is acceptable for high-volume deployments where bandwidth is a constraint.

Configure your dialer's codec preference to match. Most dialer platforms default to G.711 for outbound already. If your current carrier has required G.729 due to their infrastructure limitations, switching to G.711 over UnlimCall may produce a noticeable improvement in call audio.