It IS a trunk with multiple DID's and it IS registered with the provider.
 
Setting the "extension" parameter however does not change anything. It's like the invite request is not "matched" with the gateway config and the option is therefore not applied.

It's a pretty simple config:

<include>
    <gateway name="dev_tx1_13">
        <param name="proxy" value="x.x.x.x"/>
        <param name="ping" value="10"/>
        <param name="register" value="true"/>
        <param name="extension" value="auto_to_user"/>
        <param name="username" value="NCdWXsdSsaF2B"/>
        <param name="password" value="<censored>"/>
    </gateway>
</include>

I will contact the provider if the way they are doing it is non standard...

/Allan


On Wed, Sep 7, 2016 at 5:56 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com> wrote:

I'd ask provider to make a change and send destination number in SIP URI like most others do.

Especially if later you get different numbers from a different provider, you might benefit from having it all standard from day 1.


    
On 16-09-07 11:38 AM, Allan Kristensen
      wrote:

    

    

      
Hello,
        


        

        
I've just setup a sip trunk with a new provider, however
          there are some issues receiving calls.

        


        

        
The INVITE looks like this:

        


        

        
        
   ....

        

          
   From:
              <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203

          
        

        
   ...

        


        

        
The invite has my account name, instead of the destination
          number and the real destination number is in the "To:" field
          (which it normally is I guess ;-)

        

        


        

        
Then Freeswitch does this of cause:

        


        

        
[INFO]
            mod_dialplan_xml.c:637 Processing
            +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context
            inbound

          

        


        

        
That is undesirable, so Is there any way for Freeswitch to
          grab the destination number from the to field instead of the
          INVITE string ?

        
I could make a special sip profile for that provider with
          it's own dialplan and then set destination_number from the
          "sip_to_user" variable and transfer to the real dialplan, but
          it just seems so messy...

        
Or is it my provider that is doing it wrong?

        

        


        

        
Best regards,

        
  Allan

      

      

      

      

      
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