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How to Set Up a SIP Dialer: Step-by-Step Guide

Updated 2026-09-30 · By the ZuteSIP team (ZuteSIP Inc.)

Short answer: To set up a SIP dialer, you need three things: a phone number with SIP credentials, a SIP app or PBX, and the server details. With ZuteSIP, buy a real number, copy its SIP username and password from the dashboard, enter sip.zutesipdialer.com on port 5060 (or 5061 for TLS), choose G.711 codecs, and register.

A SIP dialer is any app that places and receives calls over the Session Initiation Protocol instead of a SIM card. This guide walks through the full SIP dialer setup, from choosing a number to confirming two-way audio, and it works the same whether you use a desktop softphone or a PBX.

What you need before you start

SIP separates the phone number from the device. Your number lives with a provider, and any SIP-compliant app can log in to it with a username and password, much like an email client logs in to a mailbox. The call signalling follows the SIP standard (IETF RFC 3261), so the same credentials work in Zoiper, Linphone, MicroSIP, Bria, Groundwire or a PBX such as Asterisk, FreePBX, 3CX or FusionPBX.

Before you open any app, gather the items below. Having them in one place avoids most of the typos that cause registration errors later.

  • A real phone number with SIP access. Every ZuteSIP number (mobile, local or toll-free, in 181 countries) includes its own SIP credentials.
  • The SIP username and SIP password for that number, shown on the number's page in the dashboard.
  • The server address: sip.zutesipdialer.com.
  • A SIP app or PBX. Any SIP-compliant client works; you do not need anything from us to install.
  • A headset or microphone, and a network that allows outbound UDP or TCP on port 5060, or TLS on port 5061.

ZuteSIP settings at a glance

These are the only values you need to enter. Field names vary slightly between apps (for example "Domain", "Registrar" or "Proxy" for the server), but the meaning is the same.

SettingValueNotes
SIP server / domainsip.zutesipdialer.comUse as registrar, domain and outbound proxy
Username / auth ID<SIP_USERNAME>Per number, from the number's page
Password<SIP_PASSWORD>Per number; copy and paste, do not retype
Transport and portUDP or TCP 5060, TLS 5061Try UDP first; switch to TCP or TLS if a network blocks it
CodecsG.711 µ-law (PCMU), G.711 A-law (PCMA)Disable other codecs or move them below G.711
AuthenticationDigestStandard SIP digest auth; no extra setting in most apps
DTMFRFC 2833 (RTP events) or SIP INFORFC 2833 is the usual default

Step-by-step SIP dialer setup

  1. Create an account at ZuteSIP with just an email address and choose a number in the country you need. Most numbers activate within minutes of payment.
  2. Open the number's page in the dashboard and copy the SIP username and SIP password.
  3. Install a SIP app, or open your PBX admin panel, and choose to add a new SIP account (sometimes called "SIP trunk", "VoIP account" or "peer").
  4. Enter sip.zutesipdialer.com as the server or domain, then paste the username and password. If the app has separate "username" and "auth username" fields, put the same value in both.
  5. Set transport to UDP and port 5060. If your network is restrictive, choose TCP on 5060 or TLS on 5061 instead.
  6. In the codec list, enable G.711 µ-law (PCMU) and G.711 A-law (PCMA) and place them at the top.
  7. Set DTMF to RFC 2833 (sometimes shown as RFC 4733 or "RTP events"). Use SIP INFO only if an automated menu does not react to key presses.
  8. Save. The account status should change to "Registered", "Online" or a green dot within a few seconds.
  9. Place a test call to another phone, then call your ZuteSIP number from a different phone to check incoming calls and two-way audio.

Example configuration (Asterisk PJSIP)

If you run your own PBX, the same values go into a trunk definition. This minimal Asterisk PJSIP example registers one number; FreePBX, 3CX and FusionPBX expose identical fields in their trunk screens. Replace the placeholders with your own values.

[zutesip-reg]
type=registration
transport=transport-udp
outbound_auth=zutesip-auth
server_uri=sip:sip.zutesipdialer.com:5060
client_uri=sip:<SIP_USERNAME>@sip.zutesipdialer.com

[zutesip-auth]
type=auth
auth_type=userpass
username=<SIP_USERNAME>
password=<SIP_PASSWORD>

[zutesip-aor]
type=aor
contact=sip:sip.zutesipdialer.com:5060

[zutesip]
type=endpoint
context=from-zutesip
disallow=all
allow=ulaw,alaw
dtmf_mode=rfc4733
outbound_auth=zutesip-auth
aors=zutesip-aor
from_user=<SIP_USERNAME>

[zutesip-identify]
type=identify
endpoint=zutesip
match=sip.zutesipdialer.com

Choosing the right app for your setup

For one person on one computer, a desktop softphone is the fastest route: MicroSIP on Windows, or Zoiper or Linphone on Windows, macOS and Linux. For a phone, Zoiper, Linphone, Bria and Groundwire all offer mobile SIP clients from their vendors.

For a team, register the number on a PBX (Asterisk, FreePBX, 3CX or FusionPBX) and route calls to extensions. Contact-centre software such as VICIdial connects the same way, as a SIP trunk. If you would rather install nothing, the browser web dialer in the ZuteSIP dashboard lets you call with any of your numbers as caller ID straight away.

Test your setup properly

A green "Registered" status only proves that the username and password were accepted. Audio travels on a separate path (RTP), so always test a real call in both directions before relying on the line.

  • Outgoing: call a mobile you own, speak from both ends, and confirm each side hears the other.
  • Incoming: call your ZuteSIP number from another phone and confirm the app rings.
  • Keypad: call an automated menu (a bank or airline line) and press a digit to check DTMF.
  • Duration: stay on a call for at least 60 seconds. Calls that drop at around 30 seconds usually point to a NAT or firewall issue.

What is included with every number

Each ZuteSIP number comes with unlimited incoming calls and SMS, 1,000 outgoing minutes and 500 outgoing SMS per month, call forwarding to any phone, the web dialer, an SMS inbox and call logs. Pricing starts from $1.80 per month, billed yearly ($21.60 per year), the same for every number type in every country, with no setup fee or contract.

Payment is in USDT or USDC only. If a number does not work for your purpose, the 7-day replacement guarantee gives you a free new number in the same country, and unused numbers are covered by a 7-day money-back guarantee.

Included on every plan

  • Unlimited incoming SMS
  • Unlimited incoming calls
  • 500 outgoing SMS per month
  • 1,000 outgoing minutes per month
  • SIP credentials, call forwarding and web dialer
  • Keep your number as long as you want
  • Works with WhatsApp, Facebook, Google, Telegram and thousands more
  • 7-day replacement guarantee — free new number if it doesn't work for you
  • 7-day money-back guarantee if the number is unused

Numbers from $1.80 per month, billed yearly ($21.60 per year). Pay with USDT or USDC.

Frequently asked questions

What is the SIP server address for ZuteSIP?

Use sip.zutesipdialer.com as the server, domain and outbound proxy. The port is 5060 for UDP or TCP and 5061 for TLS.

Where do I find my SIP username and password?

They are shown on each number's page in the dashboard. Every number has its own username and password, so a second number needs a second SIP account in your app.

Which codec should I choose?

Choose G.711 µ-law (PCMU) and G.711 A-law (PCMA). Those are the codecs supported, so put them at the top of the list and disable or lower the rest.

Can I use the same number on a PC and a phone?

Yes, you can enter the same SIP credentials in more than one SIP-compliant app. Test incoming calls on each device, because ringing behaviour depends on the app.

Do I need to install anything to make calls?

No. The browser web dialer in the dashboard places calls with any of your numbers as caller ID, so an app is optional.

Can I call emergency services over SIP?

No. Emergency calling is not supported, so always keep a regular mobile or landline available for emergencies.

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