Hello I want to follow up on this example from YEARS ago. I had tried using the variable "destination_number" but that didn't work, and I figured that it was because the To: header doesn't have the destination_number but has just the URI, so I thought I'd use sip_to_user instead.

We have calls coming in with the following info in the INVITE

From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1@67.1x.xxx.xxx:5060;user=phone>;tag=SDru6fc01-gK0c10a887.

To: <sip:011390NNNNNNNNNN;phone-context=+39@xxx.xxx.xxx.xxx:5060;user=phone>.

(N and x are obviously being masked for privacy)

I use this info in the dialplan like so

<include>

  <extension name="DIDtest">

    <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$" break="never">

        <action application="set" data="effective_caller_id_number=$2"/>

        <action application="set" data="effective_caller_id_name=$2"/>

    </condition>

    <condition field="${sip_to_user}" expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">

        <action application="set" data="continue_on_fail=false"/>

        <action application="set" data="hangup_after_bridge=true"/>

        <action application="set" data="domain_name=$${domain}"/>

        <action application="set" data="bypass_media=true"/>

        <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>

    </condition>

  </extension>

</include>

However, the calls aren't passing the condition in this dialplan and thus not being forwarded to "blade2" server. In fact, even the 011 is not being stripped off. 

What am I doing wrong?