I have set up a domain and calls between extentions work 100%. Same for outbound calls.
I have used Detinations to set up the inbound routing as per the documentation. But my inboud calls do not ork. Below is my inbound route and the result when using sngrep. Any help would be appreciated.
I am a bit of a noob with FusionPBX myself, though I have some experience with Asterisk systems (so not a *total* noob with SIP PBX systems, or rather Asterisk/FreePBX specifically). Hopefully i might be able to help you, or at least give you an "a-ha!" moment?
Is the sngrep screenshot from a run on your FusionPBX system or from the system you are calling from? I see the "User-Agent" refers to Asterisk, so I would imagine you ran it from the system you are calling through? What does sngrep show when you run it on your FusionPBX?
You are not sending a DID number in the TO field, there is nowhere to route that call to. The problem there does not lie with FusionPBX, its the box that is sending calls to it.
Hi Guys. Thanks for the response.
The sngrep shot is from my fusion. The call is being routed through my asterisk.
I ended up destroying the vm and reloading fusion. Now it tells me 480 temporarily unavailable. I have tried setting the destination (action) to extension, then ring group, then voice mail. no change.
Same as last time, where is ANY reference in your INVITE to the number being dialled. If you do not tell fusionpbx the number you are dialling, how can it possibly route that number?
We use Asterisk with an A@Billing front end. I have configured inbound DiD exactly as for all my other clients. Could this be an incompatibility between A2billing and Fusion?
Finaly I have found the issue. Thanks to DigitalDaz. In my Asterisk DID I need to not only specifi the Sipaccount, like I do for single handsets, but the called number aswell. eg: Sip.135465686489/032XXXXXXX