Is this site safe?
RegistrationRegistration is required to receive incoming calls. To do this, you must specify the SIP server. If you leave the SIP server empty, you can make calls but not be able to receive. Username, login, password and domain are also used in registration. After successful registration, you will see a green icon and the status "Online" in statusbar.
"Service unavailable", "bad gateway" or similar error.
Make sure you have entered correct "SIP server", "SIP proxy" (if needed), "Transport". Try other trasnport UDP/TCP/TLS. You can also try spoofing the user agent string in the ini file.
Error: "Forbidden", "Incorrect password" or similar.
Check your SIP server, domain, username, password. The proxy and login are often empty, but you must specify them if required by your SIP provider. Some SIP providers require that you enable the STUN server if your PC does not have a public IP address.
"Internal server error" or similar error. "SIP proxy" is not empty.
Try to add ";hide" suffix to SIP proxy, example "sipproxy.host.com;hide". This can help when SIP service configured not the best way.
CallsWhen trying to make a call a message is showed:
Error: "Unable to open sound device: Undefined external error. Error #450001" (after Windows 10 update 1803).
Fix microphone permission in the Windows settings (Windows Settings => Privacy => Microphone).
Error: "Unable to find default audio device".
Speakers and microphone both are required. To make calls you must have input and output sound device in your system. Same for RDP connections.
Error: "An invalid Parameter was passed to a system function".
Allow access to the microphone in Kaspersky Anti-virus settings.
Error: "End of file".
Change Transport to UDP.
Can't make or end a call
No sound or one directional sound
This may happen if you use one or more routers (with NAT) on the way to the PBX, or if your computer has multiple network connections.
If possible, you should configure your PBX to support NAT. For example, for Asterisk you must add "nat = auto_force_rport,auto_comedia" to the sip.conf file.
If you can't change PBX configuration, you can try to enable "Allow IP rewrite" feature, that will do that work on the softphone side and if possible disable "SIP ALG" in the router/routers settings. "SIP ALG" may interfere with the correct rewriting of IP.
The following actions may help you too:
Calls getting dropped
Long initialization time when making callsIf there is a big delay when you make a call, try to switch from TCP+UDP to UDP transport. Often SIP server does not support TCP.
Application crash or restart when making video calls.
Update your video card driver. Make sure hardware acceleration is not broken.
Green video window.
This issue is similar to the "one directional sound" problem. The video stream does not reach the softphone from the server, most likely due to the wrong network route, NAT, or firewall.
Other help pages
Before request our help please read all things above.
Report bugs and compatibility issues here
NOTICE. We can not guaranty fast answer. Your question will be queued, may be on long time.