Connect

SIP trunk connection details

Everything your telecom team needs to point a PBX or SIP trunk at Quincer. This page is the current source of truth — if a PDF, an email or a ticket disagrees with it, this page is right.

Last reviewed: 29 July 2026. Network details can change. Link to this page rather than copying the values into your own runbook.

Where to send SIP

Setting	Value
Hostname	`sip.quincer.com`
Port	`5060` (UDP, TCP) · `5061` (TLS)
Authentication	IP allowlist, or SIP digest — your choice
Audio codec	G.711 μ-law (PCMU), listed first
Packetisation	20 ms
DTMF	RFC 2833 / `telephone-event`
Called number	E.164 with `+`, e.g. `+14155550123`
Direction	Inbound to Quincer. Outbound origination is not offered today.

Point a SIP trunk, route pattern or dial peer at sip.quincer.com. We identify each call by the number dialled, so one trunk can carry many numbers, each routed to a different AI agent.

SRV records

If your platform supports SRV lookups, you can use them instead of the hostname and port. Clients that ignore SRV fall back to the A record on 5060, so either approach works.

_sip._udp.quincer.com.  SRV  10 100 5060 sip.quincer.com.
_sip._tcp.quincer.com.  SRV  10 100 5060 sip.quincer.com.
_sips._tcp.quincer.com. SRV  10 100 5061 sip.quincer.com.

The benefit is future-proofing rather than day-one setup: SRV lets us move or add capacity without you editing anything.

Firewall

All traffic is outbound from your network to ours. Nothing needs opening inbound from the internet.

The rule most often missed. Signalling on 5060 is the easy half. Audio is RTP on a separate UDP range and may not come from the same address as the signalling. A firewall that permits 5060 and nothing else produces a call that connects and then has no sound — the most common failure in this integration, and it looks like a broken AI rather than a firewall rule.

Ask us for the current media ranges before you build your rules. We would rather hand you today's values than have you discover a change mid-outage.

What we need from you

Send these through Support → Connect a SIP trunk and we'll configure your side and reply on the ticket.

Your SIP signalling IP addresses — individually, not as ranges.
Your RTP / media IP addresses, if they differ from the signalling ones. Many enterprise SBCs use a separate media block; if yours does and we don't know, calls connect silently.
The phone numbers the AI should answer, in E.164.
Your PBX or SBC make and version — it tells us which SIP dialect to expect.
Concurrent calls at your busiest. A rough number is fine; it sizes the connection.
Transport and authentication — UDP, TCP or TLS; IP allowlist or digest.

Testing

We confirm your numbers are registered and your trunk is live.
Open Integrations → Telephony, find the number and choose Check voice. For a SIP number this reports what we can see from our side: that the number is bound to your trunk, that the trunk is enabled, that your signalling IPs (or a SIP digest) are recorded, and whether a call has ever arrived on it. It does not ask you to set a webhook — a SIP trunk has no webhook. You point signalling at the host above; the rest is ours to configure.
Place a test call. The AI should answer within a second or two.
Speak, and confirm you hear the reply — both directions. One-way audio is a media firewall rule, not an AI problem.
Test a transfer to a human, if that is part of your flow.

If a call connects but is silent, it is nearly always one of two things: PCMU not listed first, or the RTP media range not permitted through the firewall. Check those before anything else.

Carrier setup examples

If your numbers sit with a carrier rather than a PBX, here is where the setting lives. The direction is the part everyone gets wrong — each console has two halves, and the names are counter-intuitive. You want the half where the carrier sends calls to an external SIP host, never the half where you send calls out to the PSTN.

Twilio — Elastic SIP Trunking

Use Origination. Twilio defines it as “receiving incoming traffic on your Twilio numbers to deliver calls to your communications infrastructure”. Termination is the opposite direction and owns the Credential Lists and IP Access Control Lists — if you find yourself adding our IP to an ACL, you are in the wrong half and no call will ever reach us.

Elastic SIP Trunking → Trunks → create a trunk.
On the trunk's Origination tab, add an Origination SIP URI: sip:sip.quincer.com:5060;transport=udp. Priority and Weight both default to 10, which is fine.
For TLS use sip:sip.quincer.com;transport=tls and turn on Secure Trunking in the trunk's General settings. The two must agree — with Secure Trunking on, unencrypted calls are rejected. Use the sip: scheme with transport=tls; sips: is not supported.
Associate your numbers with the trunk, or set each number's Voice handling to SIP Trunking and pick the trunk.

Leave the user part of the URI empty. Twilio fills it with the number that was dialled, which is how we route to the right AI agent. Set a custom user part and the same URI is used for every number on the trunk, so they all resolve to whichever one we see first.

Telnyx

Create an FQDN connection and configure its Inbound section. Ignore the Outbound tab entirely — Outbound Voice Profiles exist so your equipment can dial out through Telnyx, which is not what we are doing. Assigning a number to this connection changes inbound routing for that number only; your outbound calling is untouched.

Voice → SIP Trunking → Create SIP Connection, type FQDN.
In the INBOUND section, Add FQDN: sip.quincer.com, port 5060 (or 5061 for TLS). Set the transport to match the port.
Set Number Format (DNIS) and Number Format (ANI). See the note below.
Under Offered Audio Codecs, put PCMU / G711U first and disable G722, G729 and Opus.
Advanced Settings → DTMF Types → RFC 2833.
Assign the number: My Numbers → the SIP Connection/App column → pick your connection.
Tell us the SIP Region you chose, so we allowlist the right signalling addresses.

Codec order is the one that bites. Telnyx sends the SDP offer on inbound calls and orders it by your preference list — its own worked example puts G722 ahead of PCMU. Get this wrong and the call connects with no audio.

Number format: E.164 with or without the leading + both work — we normalise. National (10 digits) does not, because the country code is gone and we cannot infer it. The same applies to the ANI format if you want caller ID to be usable.

Raise the timers before testing. Telnyx's No Ringback Timeout and No Answer Timeout both default to 5 seconds, which can tear down a call before a cold start answers.

Bandwidth

Use the Origination half. Bandwidth uses carrier-side vocabulary, so “Termination hosts” is where you send calls out to Bandwidth — not where Bandwidth sends calls to you. Putting our hostname there is the single most common mistake on this platform.

Check which console you are on first, because the setting lives in two different places. If the Bandwidth App shows Service Management, you are on the Universal Platform; otherwise you are on classic Locations. Editing the wrong one changes nothing at all.

Universal Platform: Service Management → Voice Configuration → create a package → add a Route to Domain Name endpoint of sip.quincer.com. Then Numbers → My Numbers → Actions → Voice configuration → select the package.
Classic Locations: Account → Locations → your location → Origination Settings → Voice IP addresses / DNS hosts.

Bandwidth is UDP only. Their guides state SIP and audio must be delivered over UDP, with TCP unsupported — so the TLS option above does not apply here. They also cap packets at 1350 bytes.

Media needs UDP 1024–65534 open, which is far wider than the usual RTP range. A firewall rule written for 10000–20000 will clip audio.

Authentication is trusted-IP only — there is no digest option, and Bandwidth runs a mated SBC pair, so both of their signalling addresses must be allowlisted on our side. Bandwidth does not publish those addresses; your implementation specialist provides them. Send them to us.

Good news on Bandwidth: G.711 μ-law is already first in their codec preference, numbers are sent as +E.164 by default, and RFC 2833 DTMF is supported — so those three need no action.

Using Mitel, Cisco or Avaya? Those are on our list — open a ticket and we will walk your team through it directly in the meantime.

Good to know

We route on numbers, not extensions. If your PBX presents internal extensions rather than full DIDs, tell us — it changes the setup and we will walk you through it.
Forwarding between your own numbers works. If one of your numbers forwards to another, we honour the original so the right agent answers. Forwarding to a number belonging to a different organisation is refused by design.
Adding numbers later is quick — send us the list. No change on your side as long as they arrive over the same trunk.
Tell us before you change a signalling IP, not after. Calls from an address we do not know are rejected.

Already using a cloud phone system?

If you are on RingCentral, Teams Phone or a similar platform, you have two options. The quick one is to forward a number, queue or auto-receptionist to a Quincer number — live the same week, no trunk required. The deeper one is a carrier-level integration such as RingCentral's Bring Your Own Carrier, which puts the AI in front of your platform. Open a ticket and we will help you choose.