Browser calling and SIP access for one number
Choose the answering setup that fits your work.
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.
CallGlobal includes a browser calling interface alongside SIP and forwarding settings. A working provider connection is required; number availability and activation should be confirmed with support before funding.
Service facts before you choose
| Monthly equivalent | $1.80 for each line |
|---|---|
| Annual payment | $21.60 for the year; no setup charge |
| Subscription allowances | Unmetered inbound SMS/calls; 500 outbound texts; 1,000 outgoing minutes monthly |
| Catalog destinations | 213 |
| Line categories | Geographic, mobile and toll-free |
| External app eligibility | Decided by each platform; not guaranteed |
| Wallet funding | USDT or USDC; use the network offered for your current request |
| First-week remedies | Up to 2 eligible replacements; refund for a wholly unused line within 7 days |
| Connection status | Availability confirmation required before activation |
| Payment handling | Exact requested amount required; crypto payments need review |
Your connection options
Calling in your workspace
The browser dialer places calling controls alongside your account. Confirm activation and calling availability with support before use.
Your preferred SIP client
Configure a compatible softphone such as Zoiper, Linphone, MicroSIP, Bria or Groundwire using the confirmed settings supplied for your activated line.
Business phone systems
Use SIP configuration with supported PBX software, including Asterisk, FreePBX, 3CX, FusionPBX and VICIdial; validate routing before customer use.
An alternate destination
Forwarding controls let you define another answering phone. Confirm the destination and routing capability with the connected provider.
Protocol reference
| SIP server | Use the confirmed host supplied for your line in the dashboard |
|---|---|
| Transport | UDP / TCP 5060 · TLS 5061 |
| Codecs | G.711 µ-law / A-law |
| Authentication | Username + password, per number |
| DTMF | RFC 2833 / SIP INFO |
Prepare a compatible client
- Create a SIP profile in the client you intend to use.
- Enter confirmed server and account credentials for your assigned line.
- Verify inbound routing, outbound permissions and caller identification before customer use.
Prepare the browser dialer 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
- 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 |
Configuration reference 1
server: <SIP_SERVER_FROM_DASHBOARD>
username: <SIP_USERNAME>
password: <SIP_PASSWORD>
port: 5060 (UDP/TCP) or 5061 (TLS)
codecs: PCMU, PCMA
dtmf: rfc2833 (or SIP INFO)
caller ID: <YOUR_NUMBER>Connection field reference 2
| Setting | Value |
|---|---|
| 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 port | UDP or TCP 5060, TLS 5061 |
| Codecs | G.711 µ-law (PCMU) and A-law (PCMA) |
| Authentication | SIP digest, per-number username + password |
| 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.
Connection questions
Which credentials does the browser dialer 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.
Client reference library: Zoiper, MicroSIP, Linphone, Bria, Groundwire, 3CX, FreePBX, Asterisk, FusionPBX, VICIdial.
Understand SIP access · Explore connection guides · Review the subscription
The subscription in detail
- No monthly cap on inbound SMS or calls
- Outbound allowance: 500 SMS and 1,000 minutes each month
- SIP settings, forwarding controls and browser calling interface
- Renewable access to your selected line
- External account verification depends on the receiving platform
- First 7 days: up to 2 eligible no-cost replacements
- Refund eligibility for a line with no SMS or call usage within 7 days
The annual-plan equivalent starts at $1.80 per month, totaling $21.60 for the year. Wallet funding uses USDT or USDC and requires the exact displayed amount. Crypto payments are reviewed before credit.
Number activation requires availability confirmation. Please contact support before funding your account.