SIP24.CC: Complete Zoiper Setup & Custom Caller ID Guide

[sip24.pw]

SIP24.CC: Complete Zoiper Setup & Custom Caller ID Guide

SIP24.CC provides direct access to high-performance voice infrastructure with advanced caller ID spoofing and clean routing. Forget the public garbage apps and broken Telegram bots — stable operations require proper server configuration and reliable routing lines.

This guide walks you through connecting your SIP24 trunk to Zoiper and configuring custom caller ID spoofing for outbound calls.

Full Walkthrough: Connecting Zoiper on Your Phone

This guide walks you through downloading Zoiper, creating a SIP account inside the app, and linking it to your SIP24 credentials so you can start making calls. It should take about 5 minutes.

Step 1 — Install Zoiper Zoiper is a free softphone app available for both Android and iOS. Search for "Zoiper" on the Google Play Store or Apple App Store and install it, or use the direct links below:

• Android: Google Play Store
• iOS: Apple App Store

Step 2 — Open Zoiper and Allow Permissions When you first launch the app, it will ask for microphone and speaker access. Tap Allow for both — without microphone access, calls will connect but you won't be heard. Once permissions are granted, head into the app's Settings menu to begin adding your account.

Step 3 — Add a New Account Manually Go to Accounts and tap the "+" button to add a new account. Zoiper will ask if you already have a SIP account — since you're a SIP24 user, select "Yes". On the next screen, skip the provider list and tap "Manual configuration" instead — this lets you enter your SIP24 credentials directly.

Step 4 — Enter Your SIP24 Credentials You'll now be asked for a SIP Username, SIP Password, and Domain. You'll find all three in your SIP24 dashboard under SIP Account — copy each value exactly as shown there. Also make sure SRTP encryption is toggled on before saving, so your calls stay encrypted.

Step 5 — Register and Test Copy your Username, Password, and Domain into the matching fields in Zoiper, then tap Register. If everything was entered correctly, the account status will change to "OK" — that means you're connected and ready to make calls. Dial the echo test command *43 to confirm audio is working in both directions.

Configuration Syntax & Phone Control Codes

The system supports built-in control codes for handling caller ID injection, active call audio, and local recording directly through your webphone or standard SIP client:

• Caller ID Setup: Dial *8080*<number>. A 603 Declined response confirms the caller ID is set.
• Verify Active Caller ID: Dial *9000.
• In-Call Audio Playback: Dial *7002[number] to inject pre-recorded audio or intros into an active call.
• Blind Transfer: Use ##[number] for transfer without holding (supported on webphone and standard SIP clients).
• Local Recording: Dial *7301 to start and *7300 to stop local recording (client-side recording, nothing passes through our servers).
• Secure Media: DTLS/SRTP is enabled by default (*7443 to verify).

Route Status & Platform Access

Active routes (optimized):

• FR (France)
• US (United States)
• BE (Belgium)
• BR (Brazil)
• ES (Spain)

Accounts affected by recent optimization adjustments have been credited with 10 EUR.

• Platform Access: Access your portal via sip24.cc.
• Payments: Always request fresh payment details directly inside your user panel prior to funding. Do not use legacy payment info.

Still stuck? Remember that SIP24 support covers account, billing, and server-side issues only — we can't debug third-party app settings or device-specific quirks. If Zoiper still won't register after following every step above, double-check your balance and credentials first, then reach out via Telegram Support.

 

[carder450]

quick question for austria.

calls are working fine from my asterisk setup but the caller id is sometimes coming through as private when using

i noticed i'm sending the austrian number with the local `0` in some cases and with `43` in others.

what format should i use for the cli? i'm guessing my dialplan is probably doing something wrong here.

 

[sip24.pw]

quick question for austria.

calls are working fine from my asterisk setup but the caller id is sometimes coming through as private when using

i noticed i'm sending the austrian number with the local `0` in some cases and with `43` in others.

what format should i use for the cli? i'm guessing my dialplan is probably doing something wrong here.

For Austria, send the CLI in international format.

Use:

`43XXXXXXXXX`

Do not send the local leading `0` after the country code.

Check the actual INVITE with:

`pjsip set logger on`

The important part is what Asterisk sends in the `From:` header, not only what is configured on the extension.

If the correct CLI is present in the INVITE, send us the call time and destination and we can check it from our side.

 

[carder450]

For Austria, send the CLI in international format.

Use:

`43XXXXXXXXX`

Do not send the local leading `0` after the country code.

Check the actual INVITE with:

`pjsip set logger on`

The important part is what Asterisk sends in the `From:` header, not only what is configured on the extension.

If the correct CLI is present in the INVITE, send us the call time and destination and we can check it from our side.

yeah this was my dialplan.

i was converting the destination number but forgot to do the same thing for the cli , so sometimes it was sending the local format.

changed it to 43... and tested again. caller id is showing correctly now.

thanks.

 

[carder450]

Yes — your instinct is probably right. For Austria, don’t mix the national 0 format and the international format for the outbound CLI.
Domestic Austrian format: 0664..., 01..., 0316...
International/E.164 format: +43 664..., +43 1..., +43 316...
When converting 0... → +43..., remove the leading 0.
For Asterisk/SIP, I would normally normalize the outbound caller ID to E.164,
+436641234567
rather than sometimes sending: 06641234567

and sometimes: 436641234567


Asterisk’s CALLERID() function controls the caller-ID information, but the exact format accepted/presented ultimately depends on your SIP carrier.

Also check the SIP From, P-Asserted-Identity/P-Preferred-Identity, and the carrier’s required caller-ID format. Some providers specifically require the + E.164 form. Good dialplan approach: normalize the number once at the beginning of the outbound route, then use that normalized value consistently for CALLERID(num) and the SIP identity headers. If you paste your Asterisk outbound dialplan and a SIP INVITE from a call where the CLI works vs. one where it shows Private, I can point out exactly where the formatting is going wrong.

thank you chatgpt.

 

[sip24.pw]

this is a bump

 

[sip24.pw]

this is a bump