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CLIRoutes

Integration · Softphone

MicroSIP setup for CLIRoutes

MicroSIP is a lightweight open-source softphone for Windows. Most customers use it to test a route before pointing a dialer at it: place calls to your own mobile, check the caller ID that arrives, listen for audio quality and measure post-dial delay. It needs a SIP user on your account, which you can create under **Portal → Endpoints → Add user**.

Tested on
MicroSIP 3.21 (Windows 10/11)
Auth
Registration (SIP user)
Signalling
sip.example-carrier.net:5060 UDP/TCP · 5061 TLS
Codecs
G.711 a/μ-law, G.729, Opus
Use for
Testing, supervisors, low-volume calling

Before you start

  • A SIP user created in the portal with an authorised caller ID assigned.
  • Windows with a headset; avoid laptop speakers when judging audio.
  • No SIP ALG on your router (it rewrites headers and breaks registration).

Placeholders in angle brackets, such as <ACCOUNT_ID>, come from your portal. The host shown is a placeholder; use the one listed under Portal → Network.

Set up MicroSIP

  1. 01

    Add the account

    Open MicroSIP, click the arrow menu and choose Add Account. Fill in the fields as below and save.

    Account
    Account Name ...... CLIRoutes test
SIP Server ........ sip.example-carrier.net
SIP Proxy ......... (blank)
Username .......... <SIP_USER>
Domain ............ sip.example-carrier.net
Login ............. <SIP_USER>
Password .......... <SIP_PASSWORD>
Display Name ...... (blank)
Transport ......... UDP
Public Address .... Auto
  2. 02

    Set codecs

    In Settings → Codecs, enable G.711 a-law and μ-law first, then G.729. Disable video. This matches what dialers will negotiate, so test results transfer.

  3. 03

    Check registration

    The status bar should read *Online*. If it shows *Forbidden*, check the username and password; if it stays on *Connecting*, try TCP transport, which avoids most NAT problems.

  4. 04

    Place test calls

    Dial a mobile you control as country code plus number, for example 447700900123. Confirm the caller ID on the handset, note the time to first ring and listen for one-way audio, jitter or echo. Repeat on a fixed-line number in the same country.

What to check on a test call

CheckGood resultIf not
Caller ID on handsetExactly the authorised numberCheck the number assigned to the SIP user
Time to ringbackWithin the PDD guidance for that countryNote destination and time; send to routing
Audio both waysClear in both directions from the first secondSwitch to TCP; check firewall and SIP ALG
CDR in portalDuration matches your stopwatchReport possible false answer supervision

For a structured multi-day test, follow the 7-day route acceptance test plan and calculate results with the ASR / ACD calculator.

Last verified by the CLIRoutes routing team. Menus can change between releases; if a field has moved, tell us.

FAQ

MicroSIP questions

Can’t find what you need? Our routing team answers within the hour, around the clock.

Can agents use MicroSIP for production calling?

Yes for low-volume manual calling. For campaigns, connect a dialer so pacing, caller ID and recording are controlled centrally.

Why does MicroSIP show my caller ID as Anonymous on the handset?

The SIP user has no caller ID assigned, or the assigned number is not yet approved. Assign an authorised number in the portal.

Does MicroSIP support TLS?

Yes. Set Transport to TLS, change the server port to 5061 and enable SRTP under Media Encryption. Enable TLS on your account first.

Where are MicroSIP settings stored?

In microsip.ini in the installation or AppData folder. You can copy that file to set up several test machines with the same account.

Ready to connect MicroSIP?

Open an account to get your endpoint, credentials and authorised caller IDs.

Open account