Class of Service - Call Forwarding

We are using Class of Service to restrict access to outbound routes, we have a number of extensions and outgoing routes assigned to a class of service.

This is mostly working as expected however if an extension has call forwarding configured to an external number (e.g. mobile number) or an incoming ring-group is configured with an external number (suffixed with ‘#’); the call cannot be routed to the external number.

I assumed that as the extension has access to an outbound route then call forwarding should work?

Logs should tell you whats up.

Since Asterisk is a Back to Back User Agent, an extension forwarding a call would be unusual, especially when using Asterisk to forward the call. Unless you configure the Ring Group to be able to access the outbound, there’s no way for the system to send the call out. In your example, the extension isn’t making the call - the ring group is.

Thanks for the prompt responses, looking at the logs the calls are actually making it to our SIP Trunk provider and are being blocked there. I have now opened a ticket with them and asked them to investigate why there system is rejecting the call.

Those are the wrong logs to look at. That is for providing debugs on system level crashes and core dumps. The logs you want to look at are the standard full log for things like this.

By default, forwarded calls to external numbers (extension forwarding, follow me, ring groups, etc.) display the number of the original caller as caller ID.

Some trunking providers prohibit sending a number that is not yours. If that’s your case, you could route forwarded calls via a different provider, or set up the system to send your company’s main number instead.

Otherwise, your provider may require adding a Diversion header or putting the desired caller ID in P-Asserted-Identity or Remote-Party-ID. Another possibility is a format mismatch between the caller ID received from the original caller and what the provider expects for outgoing.

I think you are wrong on that one.
From the link:

If you think this is an Asterisk bug or a FreePBX dial plan generation bug then you need to get the Asterisk Logs

This link is for crashes:

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