Generic SIP (Asterisk / FreePBX / 3CX / FreeSWITCH)
Integration key: generic-sip
For customers running their own PBX (Asterisk, FreePBX, 3CX, FreeSWITCH, Mitel, NEC, others). You point a SIP trunk at our termination URI; we place outbound calls toward your PBX and receive inbound from it.
What you get
Section titled “What you get”- Outbound calls from CallingEdge to phones / extensions on your PBX
- Inbound calls from your PBX answered by CallingEdge
- Blind transfer via SIP REFER
- Real-time call lifecycle webhooks
- Recording, DTMF capture
What we need from you
Section titled “What we need from you”| Item | What it is |
|---|---|
| PBX SIP endpoint | The FQDN / IP and port to point at |
| Trunk credentials | Username / password if your PBX requires SIP auth |
| Inbound routing rule | A rule on your PBX that forwards inbound DIDs over the trunk to us |
| Allowlist | Our SIP source IPs added to your PBX firewall |
| Webhook URL | Where you want us to POST normalized events |
Setup at a glance
Section titled “Setup at a glance”PBX-specific examples:
- Asterisk / FreePBX: add a
chan_pjsipendpoint targeting our termination URI; create an inbound route that bridges matching DIDs to the trunk. - 3CX: Settings > SIP Trunks > Add > Generic; outbound proxy = our termination URI; inbound rules per DID.
- FreeSWITCH: gateway profile in
sip_profiles/external/, dialplan rule to bridge tosofia/gateway/callingedge/<dn>.
Send us the endpoint and (if needed) trunk credentials through the secure intake. We allowlist your PBX and confirm health.
Limitations
Section titled “Limitations”- No wrap-up code surface (generic SIP trunks do not carry dispositions).
- Attended transfer not supported on this path.