Darwvin Softphone setup guide

How to get from download to your first call on Android or Linux. You need a SIP account from your PBX or provider; the app does not include a phone number or a PBX. If a step does not work on your setup, ask us before you buy.

What you need

SIP account: User name, password and server (and port or transport if not the default) from your PBX or provider
Android: Android 9 or later
Linux: x86-64: Ubuntu 22.04 or newer (.deb), other distributions (AppImage)
Darwvin account: Free, with your email; each device signs in
Network: The PBX reachable on its SIP port (usually 5060 UDP or TCP, 5061 for TLS) and its RTP port range
Codecs: At least one of Opus, G.722 or G.711 (A-law or µ-law) allowed on the extension

Installation

1. Install the app
Download the Android APK or the Linux package from the product page. On Android, allow installs from your browser when it asks. On Debian or Ubuntu, install the .deb; on other distributions, make the AppImage executable and run it.
2. Create your free Darwvin account
Enter your email and a password, then verify the email with the link or the 6-digit code. Verifying starts the 7-day trial of every Pro feature; afterwards the app continues on Free.
3. Add your SIP account
On Android tap Add Account, then Manual SIP Setup; on Linux open Settings, then Accounts. Fill in the user name, the password and the server. Leave Transport on Auto unless your PBX needs a specific one, and turn on Use TLS (recommended) when your PBX offers SIP over TLS. If your PBX is Asterisk with PJSIP and the phone is behind NAT, add the NAT options below to the extension's endpoint section (the options are explained in the Asterisk documentation: https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Configuring-res_pjsip-to-work-through-NAT/).
pjsip.conf: NAT options to add to an existing endpoint (not a complete extension)
```
[1001]
type=endpoint
; ...your existing lines...
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
allow=!all,g722,ulaw,alaw
```
4. Read the connection check
The app checks the account straight away. Everything looks good! means you can call; We couldn't connect means the registration failed, and the account details show why. The troubleshooting list below covers the usual SIP status codes.
5. Make a test call
Call your PBX's echo-test extension if it has one, or a colleague. On Android, keep Stay connected in background on and tap Allow background and Allow call screen when the app offers them, so calls can reach you while the app is closed.

Security notes

The TLS server certificate is always checked; the check cannot be turned off.
SRTP encrypts the audio when your PBX offers it.
SIP passwords are kept in the Android Keystore or the system keychain, never in plain files.
Calls go straight between the app and your PBX; they do not pass through Darwvin.
Diagnostics exports are redacted to remove passwords and tokens; still check a file before you share it.

If something does not work

SIP 401 or 407, again and again: The user name, auth user or password is wrong. Re-enter the password; if your PBX uses a separate auth user, add it in the advanced fields.
SIP 403 Forbidden when registering: The PBX refuses this device: a wrong password, an IP access list or fail2ban, or a disabled extension. Check the extension and whether your network's public address is allowed.
Registered, but incoming calls do not ring: For safety the app only accepts calls from the server it registered with. If your provider sends calls from other addresses (an SBC), add them under the account's Trusted SIP sources.
No answer from the server after a few seconds: Check the server name and port, try Transport TCP or TLS if your PBX offers them, and make sure the network allows SIP to the PBX.
Could not verify the server certificate: Use the server name that is in the certificate. For a certificate from your own company CA, import it as a trusted organisation certificate. Certificate checks cannot be turned off.
One-way or no audio: Usually NAT on the PBX side: apply the PJSIP NAT settings above, and make sure the PBX's RTP port range is reachable. A call that fails with 488 means no common codec: allow Opus, G.722 or G.711.

With the SIP details at hand, most setups take a few minutes. Darwvin Softphone does not support emergency calls: use your mobile network or a landline for them.

What you need

SIP account

User name, password and server (and port or transport if not the default) from your PBX or provider

Android

Android 9 or later

Linux

x86-64: Ubuntu 22.04 or newer (.deb), other distributions (AppImage)

Darwvin account

Free, with your email; each device signs in

Network

The PBX reachable on its SIP port (usually 5060 UDP or TCP, 5061 for TLS) and its RTP port range

Codecs

At least one of Opus, G.722 or G.711 (A-law or µ-law) allowed on the extension

Installation

1. Install the app

Download the Android APK or the Linux package from the product page. On Android, allow installs from your browser when it asks. On Debian or Ubuntu, install the .deb; on other distributions, make the AppImage executable and run it.

2. Create your free Darwvin account

Enter your email and a password, then verify the email with the link or the 6-digit code. Verifying starts the 7-day trial of every Pro feature; afterwards the app continues on Free.

3. Add your SIP account

On Android tap Add Account, then Manual SIP Setup; on Linux open Settings, then Accounts. Fill in the user name, the password and the server. Leave Transport on Auto unless your PBX needs a specific one, and turn on Use TLS (recommended) when your PBX offers SIP over TLS. If your PBX is Asterisk with PJSIP and the phone is behind NAT, add the NAT options below to the extension's endpoint section (the options are explained in the Asterisk documentation: https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Configuring-res_pjsip-to-work-through-NAT/).

pjsip.conf: NAT options to add to an existing endpoint (not a complete extension)

[1001]
type=endpoint
; ...your existing lines...
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
allow=!all,g722,ulaw,alaw

4. Read the connection check

The app checks the account straight away. Everything looks good! means you can call; We couldn't connect means the registration failed, and the account details show why. The troubleshooting list below covers the usual SIP status codes.

5. Make a test call

Call your PBX's echo-test extension if it has one, or a colleague. On Android, keep Stay connected in background on and tap Allow background and Allow call screen when the app offers them, so calls can reach you while the app is closed.

Security notes

The TLS server certificate is always checked; the check cannot be turned off.

SRTP encrypts the audio when your PBX offers it.

SIP passwords are kept in the Android Keystore or the system keychain, never in plain files.

Calls go straight between the app and your PBX; they do not pass through Darwvin.

Diagnostics exports are redacted to remove passwords and tokens; still check a file before you share it.

If something does not work

SIP 401 or 407, again and again

The user name, auth user or password is wrong. Re-enter the password; if your PBX uses a separate auth user, add it in the advanced fields.

SIP 403 Forbidden when registering

The PBX refuses this device: a wrong password, an IP access list or fail2ban, or a disabled extension. Check the extension and whether your network's public address is allowed.

Registered, but incoming calls do not ring

For safety the app only accepts calls from the server it registered with. If your provider sends calls from other addresses (an SBC), add them under the account's Trusted SIP sources.

No answer from the server after a few seconds

Check the server name and port, try Transport TCP or TLS if your PBX offers them, and make sure the network allows SIP to the PBX.

Could not verify the server certificate

Use the server name that is in the certificate. For a certificate from your own company CA, import it as a trusted organisation certificate. Certificate checks cannot be turned off.

One-way or no audio

Usually NAT on the PBX side: apply the PJSIP NAT settings above, and make sure the PBX's RTP port range is reachable. A call that fails with 488 means no common codec: allow Opus, G.722 or G.711.

With the SIP details at hand, most setups take a few minutes. Darwvin Softphone does not support emergency calls: use your mobile network or a landline for them.