SIP Trunks: bring your own carrier
Connect your existing SIP carrier (NetGSM, Twilio, Telnyx, Asterisk) to Call2Me — keep your numbers and contracts, get the voice AI on top.
Updated May 6, 2026
A SIP trunk lets Call2Me handle calls on numbers that live with your existing carrier. The carrier keeps the relationship with the phone network; Call2Me handles the AI and audio.
Adding a trunk
curl -X POST https://api.call2me.app/v1/sip-trunks \
-H "Authorization: Bearer YOUR_API_KEY" \
-H "Content-Type: application/json" \
-d '{
"name": "NetGSM Production",
"sip_server": "sip.netgsm.com.tr",
"sip_username": "your_username",
"sip_password": "your_password",
"transport": "udp",
"outbound_caller_id": "+908501234567"
}'
The dashboard equivalent is Phone Numbers → SIP Trunks → New Trunk.
Inbound: routing carrier traffic to Call2Me
On your carrier side, point the number at Call2Me's SIP endpoint. The
endpoint is shown in the dashboard once the trunk is registered (e.g.
sip:inbound@sip.call2me.app:5060).
In Call2Me, bind the number to an agent:
curl -X POST https://api.call2me.app/v1/phone-numbers \
-d '{
"phone_number": "+908501234567",
"sip_trunk_id": "trunk_xyz",
"agent_id": "agent_abc123"
}'
When the carrier delivers an inbound INVITE, the platform answers with your agent.
Outbound: dialing through your trunk
Set from_number on a POST /v1/calls to a number bound to your trunk.
The platform routes the outbound INVITE through your carrier; recipients
see your number as the caller ID.
Transports & security
| Transport | When |
|---|---|
| UDP | Default, fast, most carriers |
| TCP | When middleboxes drop UDP |
| TLS | When you need encrypted signaling for compliance |
For media: SRTP is negotiated automatically when both sides support it. For full encryption end-to-end, set the trunk to TLS and ensure your carrier offers SRTP.
IP whitelisting
The platform's signaling and media IPs are listed in the dashboard (Settings → Network). Whitelist them on:
- your carrier's firewall (for inbound)
- your PBX or SBC (for outbound)
- any cloud security group between you and the carrier
These IPs are stable — we publish notices before any change.
Troubleshooting
Most SIP trunk issues are one of three things:
- Auth fails (401, 403) — username/password mismatch, or the carrier
expects a specific
Fromdomain - Calls connect but no audio — RTP port range not whitelisted; check carrier's RTP requirements
- Calls drop after 32 seconds — classic NAT/keepalive issue; switch transport to TCP or enable SIP outbound on the carrier
The dashboard's per-trunk SIP log is the fastest path to a specific cause.
What's next
- Phone Numbers — number lifecycle on top of a trunk
- Calls — placing outbound calls via your trunk
- Voice — codec and audio quality tuning
Frequently asked
Q.Why use a SIP trunk instead of buying a number?
Three reasons: you already own numbers you can't easily move, you have a regulatory or carrier requirement, or you need volume pricing only your existing carrier offers. Otherwise, buying through Call2Me is faster.
Q.Which SIP providers work?
Anything that speaks SIP over UDP/TCP/TLS. Tested with NetGSM, Twilio, Telnyx, Vonage, and self-hosted Asterisk/FreeSWITCH. Codecs: PCMU, PCMA, OPUS.
Q.Do I need to whitelist Call2Me's SIP IPs?
Yes. The platform sends signaling and media from a documented set of IPs visible in the dashboard. Whitelist these on your carrier or PBX firewall.
Q.What about TLS and SRTP for compliance?
Both supported. Set transport to TLS in the trunk config; the platform negotiates SRTP automatically when the carrier offers it.