Outbound calls ok, inbound calls ok, inbound call rerouted to outbound call 'all circuits busy'

Direct outbound calls and outbound calls via a misc destination are working fine as are inbound calls.
When an inbound call is routed to a misc destination we get ‘all circuits busy’.

My install is FreePBX

In the hope of clarifying things more …

014xxxxxx=phone number connected to my trunk
047xxxxxxx=a cell phone
050xxxxxx=a landline

Dialing 047xxxxxxx on extension = ok.
Dialing #047(=misc application to misc destination to 047xxxxxxx) on extension = ok.

047xxxxxxx to 014xxxxxx(=inbound route to extension) = ok.

050xxxxxx to 014xxxxxx(=inbound route to misc destination to 047xxxxxxx) = all circuits busy.

Did I miss some configuration? How can I troubleshoot this further?

This is probably the same caller ID problem discussed in your other thread. If so, you either need to fix the issue of sending numbers that are not yours, or do the reroute with a Ring Group or other means where you can force the outbound caller ID to be a company number.

Another possibility is a limitation on the number of simultaneous calls permitted on the trunk, imposed by the trunking provider or by configuration set in the PBX.

Finally, it’s possible that the inbound trunking provider requires (at least) a Ringing or Progress signal within e.g 5 seconds, and the setup time for the outbound leg is longer than that. Using a Ring Group would avoid that.

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This is another FreePBX install. In the install of my other thread, I can reroute inbound to outbound calls, only the original CID is lost.

The SIP provider of this install is OVH. I have a trunk with 5 channels > https://www.ovhtelecom.fr/telephonie/sip-trunk/. Maximum channels of trunk configuration is also set to 5.

But this will be the problem … I tried to make a second outbound call from another extension which indeed fails.

I have 2 older FreePBX installs that also use an OVH trunk and that have no problem to make simultaneous calls. I’m googling now …

Look at the Asterisk log for a failed call. If "Called " (without the quotes) appears, the call was sent to the trunk and rejected by the provider. If not, FreePBX rejected it, possibly by counting channels incorrectly.

If the provider rejected the call, at the Asterisk command prompt type
pjsip set logger on
sip set debug on
according to trunk type. Make another failed call – the response to INVITE should give a clue as to why the call was rejected.

Call was sent and rejected.

SIP/2.0 403 Too many simultaneous sessions

A SIP channel is equal to a session or concurrent call, right?

A SIP Channel is half a call.

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