Hello

With or without the extension parameter, it's exactly the same.

Thanks

On 01/05/2015 20:04, Stanislav Sinyagin wrote:

Remove the extension parameter and see if it helps.

On May 1, 2015 6:11 PM, "Tanguy" <phenix@vfemail.net> wrote:
Hello,

My provider did not send correct DID number in the INVITE packet but i can use "To" argument

INVITE sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429 SIP/2.0.
Call-ID: 25016-VB-188fd96b-526e3dbd4@sip.ovh.fr.
Contact: <sip:10.7.1.60:5060>.
Content-Type: application/sdp.
CSeq: 403749831 INVITE.
From: "0967212xxx" <sip:0967212xxx@sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
Max-Forwards: 27.
Record-Route: <sip:91.121.129.20:5060;lr>.
To: <sip:0557590xxx@10.7.1.60;user=phone>.
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.

Using asterisk i can bypass the issue using something like  exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am unable to do the same under freeswitch.

My trunk configuration seems correct, as you can see i used  auto_to_user, but the destination number remains 0033972480xxx when i call 0557590xxx.

2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 0967212xxx <0967212xxx>->0033972480xxx in context public


<include>
    <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
      <param name="username" value="0033972480xxx"/>
      <param name="password" value="xxxxxxx"/>
      <param name="proxy" value="sip.ovh.fr"/>
      <param name="expire-seconds" value="800"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="30"/>
      <param name="extension" value="auto_to_user"/>
      <param name="context" value="public"/>
    </gateway>
</include>

I tried to edit my inbound dialplan manually, it works using  <condition field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way to do this because i will also use telcos with normal invite packets

I how i can copy $sip_to_header to destination for this specific trunk ?

Please note that i use fusionpbx.

Best regards, sorry for my bad English