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A SIP connection links Callers to your own telephony: a carrier (Telnyx, Plivo, CommPeak, Twilio Elastic SIP Trunking, and others), a PBX, or a contact-center platform. Your agents then place outbound calls through your trunk and answer calls your carrier routes to Callers, using your numbers and your carrier rates.

Before you start

Get these details from your carrier or PBX administrator:
  • Outbound (Callers → carrier): the SIP server address (domain or IP), port, and transport. Also how the carrier authenticates you: username and password (digest), registration, IP allowlisting, or a tech prefix.
  • Inbound (carrier → Callers): the IP addresses your carrier sends calls from.
  • Numbers: the phone numbers on the trunk, in E.164 format (for example +12025550123).
Then give your carrier the Callers address to send inbound calls to:
Send inbound calls to:
or IP address 35.245.182.192.This address only receives calls. Outbound calls from Callers come from different addresses. If your carrier authenticates outbound calls by IP, email for the addresses to allowlist.

Create a SIP connection

1

Open SIP Integration

Go to App Center, find SIP Integration under Telephony & Communication, and click +. The Add SIP window opens.
App Center search for SIP showing the SIP Integration card under Telephony & Communication with a plus button

SIP Integration in the App Center

2

Fill in the general settings

Enter a SIP Name and turn on the options your carrier requires. See General settings.
3

Configure gateways

On the right, set up at least one gateway: an Outgoing gateway to place calls and an Incoming gateway for each IP your carrier sends calls from. See SIP gateways.
Add SIP window with general settings and gateways

Add SIP: general settings and gateways

4

Add phone numbers

Click Next and enter the numbers on the trunk. You can paste several at once. You can also add numbers later from Phone numbers → Add SIP Numbers.
5

Test the connection

Open the connection again and click Test SIP. See Test your connection.
Once saved, the numbers appear on the Phone numbers page under SIP - <SIP Name>. You can assign them to any voice agent.

General settings

SIP gateways

A gateway is a SIP endpoint on your carrier’s side. Add as many as you need with Add Another SIP Gateway.
For redundancy, add your carrier’s primary and secondary servers as separate outgoing gateways, and add every signaling IP your carrier uses as an incoming gateway. Calls from IPs that aren’t listed are rejected.

Phone numbers

Add each number on the trunk in E.164 format. Callers uses them to:
  • Route inbound calls: a call to one of these numbers is answered by the incoming agent the number is assigned to.
  • Set caller ID on outbound calls: agents present the assigned number to the person they call.
To manage numbers later, open the connection and go to the Phone Numbers tab, or use Phone numbers → Add SIP Numbers. You can delete SIP numbers one at a time or in bulk from the Phone numbers page. Numbers in use by an agent are listed with the agents that use them.

Test your connection

Test SIP places a real call through your trunk and reports the SIP response, so you can confirm routing and authentication before you assign numbers to an agent.
1

Open the connection

In the App Center, open your SIP connection. Test SIP is available once the connection is saved, has at least one number, and has no unsaved changes.
2

Choose the numbers

In Call from, pick one of the connection’s numbers. In Call to, enter a phone you can answer.
3

Run the test

Click Test SIP. The result shows the SIP status. Click Try Again to run it again.
Test SIP checks outbound calling. To test inbound, call one of the trunk’s numbers after assigning it to an incoming agent.

SIP headers

Callers can pass contact data to your PBX or contact center as custom SIP headers, so the receiving system can route or screen-pop the call. Outbound calls and transfers. With Send SIP Headers on (agent settings, on by default), each contact variable is added as a custom X- header on the outbound INVITE and on transfers. The contact’s phone number is included as phoneNumber.
  • Header names are the fixed prefix X-Voxia- followed by the variable name in lowercase. Characters that aren’t allowed in header names become _. To see the exact names your PBX receives, place a test call and inspect the INVITE.
  • Each value is limited to 256 bytes, and all variable headers together to about 1 KB. Short values are kept whole. Long values are shortened, and if there are too many variables the longest are dropped. Line breaks in values are replaced with spaces.
  • Transferred calls also carry X-Parent-Call-Sid with the session ID of the original call (sessionId in the webhook payload).
For a contact with the variables firstName, accountId, and tier, the INVITE your carrier or PBX receives looks like this (headers unrelated to Callers omitted):
Callers sends the names fully lowercased (x-voxia-firstname). SIP header names are case-insensitive, and most PBXs and packet analyzers display them capitalized as above, so match on them case-insensitively. A transfer from that call adds one header:
Use X-Parent-Call-Sid to pull up the AI part of the conversation (transcript, summary) for the person taking the transfer. Turn Send SIP Headers off if your carrier rejects INVITEs with custom headers, or if the variables contain data you don’t want to leave Callers. Inbound calls. Your PBX can set variables for an incoming call by sending an X-Vars header with a JSON object:
The agent can use {{firstName}}, {{accountId}}, and {{tier}} in its script for that call. The contact’s email can also be passed this way for Cal.com bookings.

Transfers over SIP

Agents on SIP calls can transfer to a phone number or directly to a SIP URI on your PBX (for example sip:queue@pbx.example.com). See Call transfers.

SIP REFER

By default, Callers bridges transferred calls: it stays on the call, can record it, and can play a handoff message for a warm transfer. With SIP REFER, Callers instead asks your carrier to take over the call and hangs up immediately.
With SIP REFER:
  • The call isn’t recorded after the transfer.
  • Callers can’t tell whether the transfer was answered.
  • Warm transfers, handoff messages, and any post-transfer logic don’t run.
SIP REFER is set per SIP connection and applies to every transfer on it. Email to turn it on.

Multiple SIP connections

You can create as many SIP connections as you need, for example one per carrier, region, or business unit, or a backup trunk. Each connection has its own settings, gateways, and numbers. Agents use whichever numbers you assign to them.
App Center with an existing SIP connection named SIP1 under Connected and an arrow pointing to the plus button on the SIP Integration card

Add another SIP connection with the plus button

Troubleshooting

  • Your carrier must send calls to sip.callers.ai (or 35.245.182.192).
  • Add every IP your carrier sends from as an Incoming gateway. Calls from unknown IPs are rejected.
  • The dialed number must be on the connection in E.164 format and assigned to an incoming agent.
Turn on Outbound authentication and check the credentials. If your carrier requires registration, turn on Require SIP Register and set SIP Realm.
The carrier refused the call. Common causes: the carrier authenticates by IP (contact support for Callers’ outbound addresses), the caller ID isn’t authorized on the trunk (try SIP Diversion Header), the tech prefix is missing, or the destination country isn’t enabled on your carrier account.
This is usually a firewall or NAT issue for the media (RTP) stream. Make sure your carrier or PBX allows RTP from Callers, and try Send OPTIONS ping. Download the call’s PCAP from the call view and share it with your carrier or our support team.
You may be exceeding your carrier’s calls-per-second limit. Lower Calls Per Second (CPS) Limit to match it.
Turn off Send SIP Headers in the agent’s settings, or reduce the number and size of contact variables.

Telephony overview

Compare SIP with Twilio and Callers-managed numbers.

Call transfers

Transfer to numbers, SIP URIs, and human operators.

Telnyx

Step-by-step Telnyx trunk setup.

Twilio SIP

Twilio Elastic SIP Trunking and Flex.

Troubleshooting

One-way audio, registration failures, and rejected calls.

Voices and sounds

Background sounds, which play only on SIP calls.