2026-03-15
IP2IP (IP-to-IP) authentication lets you connect to SpoofGlobal's VoIP infrastructure using your server's IP address instead of a username and password. Any SIP traffic originating from your whitelisted IP is automatically accepted and routed. This is the preferred method for PBX systems, call centers, and high-volume operations.
Add a trunk pointing to SpoofGlobal's server. No authentication is needed — just set the host.
In your Asterisk sip.conf or pjsip.conf:
Create a gateway in your SIP profile pointing to our server IP. Set register to false since IP2IP doesn't require registration.
Add a SIP trunk with the server IP. Select "No registration" as the authentication type. Route outbound rules through this trunk.
Set your default caller ID in the Telegram bot. This is what recipients see when you call. You can also override the caller ID per-call by setting it in your PBX's outbound route or dial plan — whatever your PBX sends in the SIP From header will be used.
Make sure you have an active route in the bot for the countries you want to call. Without a route, calls will fail with a 404 error.
IP2IP users can override the bot-set caller ID by sending their own caller ID in the SIP INVITE. The default caller ID (set in the bot) is used as a fallback when no caller ID is sent from your PBX. This gives you full flexibility to change caller ID dynamically per-call from your PBX dialplan.
No. IP2IP requires a static IP address. If your IP changes, you'll need to update it in the bot each time. For dynamic IPs, use standard SIP authentication instead.
There is no hard limit on concurrent calls with IP2IP. Your capacity depends on your balance and your server's bandwidth.
Yes. You can have regular SIP accounts and IP2IP access on the same SpoofGlobal account. Use whichever suits your current setup.
No. IP2IP does not use SIP registration. Your server simply sends calls to our IP and they are routed automatically based on your source IP.