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Connect your preferred calling tools

Updated 2026-10-30 · Prepared by Call Global Communications Inc

Short answer: Choose the browser dialer for a ready-to-use calling interface, a softphone for a dedicated application, or a compatible PBX for your own extension and routing setup.

SIP software 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.

  • Zoiper — Explore the zoiper reference and decide how it fits your telephone workflow.
  • MicroSIP — Explore the microsip reference and decide how it fits your telephone workflow.
  • Linphone — Explore the linphone reference and decide how it fits your telephone workflow.
  • Bria — Explore the bria reference and decide how it fits your telephone workflow.
  • Groundwire — Explore the groundwire reference and decide how it fits your telephone workflow.
  • 3CX — Explore the 3cx reference and decide how it fits your telephone workflow.
  • FreePBX — Explore the freepbx reference and decide how it fits your telephone workflow.
  • Asterisk — Explore the asterisk reference and decide how it fits your telephone workflow.
  • FusionPBX — Explore the fusionpbx reference and decide how it fits your telephone workflow.
  • VICIdial — Explore the vicidial reference and decide how it fits your telephone workflow.

Prepare SIP software for the line

Choose the browser dialer for a ready-to-use calling interface, a softphone for a dedicated application, or a compatible PBX for your own extension and routing setup.

Values and checks before use

  1. Open the active number in your CallGlobal dashboard and obtain its current SIP server, username and password.
  2. Place each value in the corresponding account field; do not reuse the website login as the SIP password.
  3. Use a supported transport and audio configuration, then check whether the client reports a registered account.
  4. Inspect microphone access and audio routing separately if registration works but calls have no sound.

Supported connection details

SettingReference
SIP signalingUDP/TCP 5060; TLS 5061
Audio codecsG.711 PCMU and PCMA
Keypad eventsRFC 2833 or SIP INFO
AuthenticationSeparate username and password supplied for each line

Connection field reference 1

SettingValue
SIP server / domain / registrar<SIP_SERVER_FROM_DASHBOARD>
Username / auth ID<SIP_USERNAME> — copy the credential assigned to this line in your account
Password<SIP_PASSWORD> (from the same page)
Transport and portUDP or TCP 5060, TLS 5061
CodecsG.711 µ-law (PCMU) and A-law (PCMA)
AuthenticationSIP digest, per-number username + password
DTMFRFC 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 SIP software 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.

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