Can receive call but can`t make calls

[david55]

You’ve forgotten to include the logs:

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug

Setting up SIP and PJSIP debug

From a root shell prompt, type asterisk -r to get to the Asterisk CLI. You can then type pjsip set logger on and/or sip set debug on You won’t need both, unless you have a mix of pjsip and chan_sip extensions and trunks. Some jitter is normal and inevitable. Excessive jitter that causes choppy voice or other quality issues is not and should be tracked down, but pjsip logger and sip debug are not the right tools for that. Also, excessive jitter does not normally result in dropped calls…

Is the announcement coming from FreePBX or the provider? I seem to remember that the FreePBX message similar to this is not exactly as you quote.

Possibilities include not having an outgoing route for the number you have dialled, or that route’s not modifying the number into the form required by the provider.