SIP Connectivity

Connect your PBX, SBC or SIP trunk to Callr. This page lists everything you need to configure your side: endpoints, IP ranges to allow, the expected INVITE format, codecs, and how to read call rejections.

Transports

IP ranges to allow

Allow both the SIP signaling and the RTP media ranges of the region you connect to in your firewall.

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Webhooks and fetch

These ranges only cover SIP signaling and RTP media. HTTP requests sent by Callr (Webhooks and fetch@v3) come from the ranges listed in IP Subnets.

Sending calls to Callr

Number format

By default, numbers are in E.164 format prefixed by +, in the Request-URI, From and To:

INVITE sip:[email protected] SIP/2.0

Each trunk group of the Console (VoIP → SIP termination) can use another format:

Prefix

When the same IP address is used on several trunk groups or sub-accounts, the trunk group has a prefix. Send it before the number, followed by #, or in the X-Callr-Account-Code header:

INVITE sip:GP81PGTA#[email protected] SIP/2.0

INVITE example

A complete INVITE sent by your SBC (203.0.113.10) for a call from +33612345678 to +33199001234:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 203.0.113.10:5060;branch=z9hG4bK-524287-1
Max-Forwards: 70
From: <sip:[email protected]>;tag=8a2f6e1c
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS
User-Agent: MySBC/1.0
Content-Type: application/sdp
Content-Length: 319

v=0
o=- 3912345678 3912345678 IN IP4 203.0.113.10
s=-
c=IN IP4 203.0.113.10
t=0 0
m=audio 16384 RTP/AVP 8 0 9 18 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:111 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Codecs

Callr supports all codecs. Media is relayed end-to-end: offer the codecs you support in your SDP, in your order of preference, and the codec is negotiated with the other party. When a specific codec list is required, Callr transcodes the media to it (e.g. PCMA, G.729, Opus).

Call rejections: the X-HupReason header

When Callr rejects a call, the SIP response always carries an X-HupReason header with the reason. The SIP status code alone is generic (403, 404, 484, 488, 503, 603…): read X-HupReason to know why the call was refused.

SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.0.113.10:5060;branch=z9hG4bK-524287-1
From: <sip:[email protected]>;tag=8a2f6e1c
To: <sip:[email protected]>;tag=Hv5Q7r3BmS0ca
Call-ID: [email protected]
CSeq: 1 INVITE
X-HupReason: [DESTINATION_NOT_ALLOWED] Destination not allowed
Content-Length: 0

The value is [CODE] Message: the code is stable and can be used in your logic, the message is for humans and may change. The most common codes:

When the call failed further down the line (no answer, the destination network refused it, internal error), X-HupReason carries a plain message instead, such as Timeout [60] or Internal error.

Calling a SIP endpoint from Callr

To make a call to a SIP endpoint, use dial@v2 in a Callr Actions scenario.

Here is an example dialing to an OpenAI Realtime agent (see Connect to OpenAI for the full setup, including the webhook your server needs):

description: Dial to SIP

compat:
  version: "2025-09-01"

defaults:
  autoAnswer: false

branches:
  inbound-call:
    actions:
      - action: dial@v2
        params:
          targets:
            - number: sip:[email protected];transport=tls
              timeout: 60

Other guides: Connect to ElevenLabs, Connect to Vapi.