'All circuits are busy now' when using sip trunk

Hi everyone,

I set up a simple freepbx setup, just 2 pjsip extensions, 1 pjsip trunk and a default in and out route.
i get the message ‘all the circuits are busy now’ everytime i try to call to pstn.

Public IP address and Local network are described in Asterisk SIP Settings page.

here’s a log of the problem: https://pastebin.com/rArKx3Hh

Here is your dial attempt:

[2018-03-14 17:18:15] VERBOSE[5378][C-00000004] pbx.c: Executing [[email protected]:31] Dial("PJSIP/103-00000008", "PJSIP/3485754094 @voipvoice_test,300,T") in new stack
[2018-03-14 17:18:15] VERBOSE[5378][C-00000004] app_dial.c: Called PJSIP/3485754094 @voipvoice_test
[2018-03-14 17:18:15] VERBOSE[5378][C-00000004] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

Your provider is rejecting the dial attempt, possibly the number is not formatted correctly. Whatever the reason, they should be able to tell you why.

I don’t think my provider is the problem because if I set the same trunk into my voip phone or any softphone, it works without problem.

Here’s the part Lorne was talking about - the extra space in the dial string will invalidate it and your provider should reject it.

I looked at the pastebin i posted and the space between 4 and @ is not present. In the pastebin is [email protected] as it should be

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