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A standard SIP trunk is how most customers connect. Your phone system points a trunk at our SIP ingress, calls flow across it, and the AI answers. Nothing about your setup changes: your PBX keeps its numbers, queues, extensions, recording, and carrier.
This is the default connection mode. If you are not sure which mode you need, this is almost certainly the one. Other options exist for special cases, and they are covered on the connection modes page.

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What it is

A trunk is a persistent SIP relationship between two systems. In this case the two systems are your PBX (or carrier) and our SIP ingress at sip.voice.getsmartalex.com. Two things travel over the trunk:
  • Signalling, on ports 5060 and 5061, which sets up, controls, and tears down each call.
  • Media, the actual audio, carried as RTP on UDP ports 10000 to 60000.
When a call arrives, your PBX sends it down the trunk to our ingress, the voice engine answers and talks to the caller, and if a human is needed the call is handed back to your PBX so it rings the right extension. Your phone system stays in charge of everything it already does.

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When to use it

You run a normal PBX

3CX, Yeastar, FreePBX, Grandstream, Avaya, Mitel, or anything that speaks SIP. A standard trunk is the intended path.

You want the fastest setup

One trunk, one firewall rule, one test call. No extra hardware, no session border controller.

You keep your numbers and carrier

Your DIDs and your carrier are untouched. The AI answers on numbers you already own.

You want your existing routing to stay

Queues, ring groups, voicemail, and CDRs keep working exactly as they do today.
Reach for a different connection mode only if you have a specific constraint that a standard trunk cannot meet, such as a private peering requirement or a platform that cannot originate an outbound trunk. See connection modes for the full comparison.

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Set it up

1

Point a trunk at our ingress

In your PBX, create an outbound SIP trunk whose destination host is sip.voice.getsmartalex.com. Use the credentials from your SmartAlex account for digest authentication. Signalling uses port 5060 for UDP or TCP, and port 5061 for TLS.
2

Allow the media range on your firewall

Open outbound UDP to ports 10000 to 60000 for RTP audio, with the stateful return path allowed. This is the single most common cause of one-way audio when it is missed. The full firewall detail, including SIP ALG and session timeouts, is on the network requirements page.
3

Test a call

Route a test number, or a temporary inbound rule, to the new trunk and place a call. The AI should answer, hold a conversation, and transfer back to a human when asked. If audio is missing in one direction, revisit the media range and confirm SIP ALG is off.

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How the trunk fits together

The diagram below shows the two paths that make up the trunk: signalling on 5060 and 5061, and media on the 10000 to 60000 UDP range. Both run between your PBX and our SIP ingress. The AI handles the first touch. When a caller needs a person, the call is handed back down the trunk so your PBX rings the extension, queue, or ring group using its own rules.

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Common questions

No. The trunk is originated outbound from your PBX to our ingress. You allow outbound signalling and the media range, and stateful firewalls handle the return path automatically. Inbound rules from the internet to your PBX are not required.
Use UDP or TCP on 5060 for standard signalling, or TLS on 5061 if you want encrypted signalling. All three reach the same ingress. Start with 5060 unless your policy requires encryption in transit.
Yes. Transfers come back to your PBX as standard SIP, so your call detail records, recording, and reporting continue to work as they do today.

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Next steps

Network requirements

Firewall rules, ports, SIP ALG, and the change request template your network team needs.

Connection modes

Compare the standard trunk with the other ways to connect, and when each one applies.