Connect FreePBX to your CallGlobal line
Short answer: Create an appropriate PJSIP trunk in FreePBX using the supplied registration credentials. Configure the corresponding inbound route and outbound route; a registered trunk by itself does not select an extension.
FreePBX handles the phone interface or routing environment; CallGlobal supplies the active line and its account credentials. Keep those responsibilities separate when setting up and troubleshooting.
Prepare FreePBX for the line
Create an appropriate PJSIP trunk in FreePBX using the supplied registration credentials. Configure the corresponding inbound route and outbound route; a registered trunk by itself does not select an extension.
Values and checks before use
- Open the active number in your CallGlobal dashboard and obtain its current SIP server, username and password.
- Place each value in the corresponding account field; do not reuse the website login as the SIP password.
- Use a supported transport and audio configuration, then check whether the client reports a registered account.
- Inspect microphone access and audio routing separately if registration works but calls have no sound.
Supported connection details
| Setting | Reference |
|---|---|
| SIP signaling | UDP/TCP 5060; TLS 5061 |
| Audio codecs | G.711 PCMU and PCMA |
| Keypad events | RFC 2833 or SIP INFO |
| Authentication | Separate username and password supplied for each line |
Connection field reference 1
| Setting | Value |
|---|---|
| SIP server / registrar / domain | <SIP_SERVER_FROM_DASHBOARD> |
| Transport and port | UDP or TCP 5060; TLS 5061 |
| Username | <SIP_USERNAME> — copy the credential assigned to this line in your account |
| Password | <SIP_PASSWORD> (from the same page) |
| Authentication | Registration authenticates through Digest using the assigned username and password |
| Codecs | G.711 µ-law (PCMU) and G.711 A-law (PCMA) |
| DTMF | RFC 2833 (RTP events) or SIP INFO |
Keep the connection secure
Restrict access to the SIP password and to any PBX administration panel. Redact credentials before sharing logs, and keep the number renewed so the account remains usable.
CallGlobal does not supply campaign automation, a hosted PBX or predictive dialing. Configure application-specific behavior in the software you operate and follow the rules governing each call.
Frequently asked questions
Which credentials does FreePBX need?
Use the SIP values attached to the active number. These are separate from the email and password used to sign in to the CallGlobal website.
What if the account registers but audio fails?
Check microphone permissions, the selected audio device, codec compatibility and firewall or NAT handling. Registration and audio use different parts of the connection.
Can the browser be used instead?
Yes. The dashboard includes a web dialer, so a separate SIP client is optional for ordinary browser calling.
Continue exploring
- PBX and softphone integrations
- 3CX SIP trunk setup
- Asterisk PJSIP configuration
- FusionPBX gateway setup
- what a PBX is
- fixing one-way audio and NAT
- SIP troubleshooting
- pricing for every country
- Connect Bria to your CallGlobal line
Review these country destinations
- United States phone number (+1)
- United Kingdom phone number (+44)
- Canada phone number (+1)
- Australia phone number (+61)
- India phone number (+91)
Sources
- Sangoma Documentation: PBX GUI - Trunks Module User Guide (as of September 2026)