[Freeswitch-users] sip_to_user and destination number

[Tanguy]

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 at vfemail.net 
> <mailto:phenix at 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 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429
>     <mailto:sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429>
>     SIP/2.0.
>     Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr
>     <mailto:25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr>.
>     Contact: <sip:10.7.1.60:5060 <http://10.7.1.60:5060>>.
>     Content-Type: application/sdp.
>     CSeq: 403749831 INVITE.
>     From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
>     <mailto:sip:0967212xxx at 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 at 10.7.1.60;user=phone>
>     <mailto:sip:0557590xxx at 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,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but
>     i am unable to do the same under freeswitch.
>
>     My trunk configuration seems correct, as you can see i used 
>     auto_to_user, but the destination number remains 0033972480xxx
>     when i call 0557590xxx.
>
>     2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635
>     Processing 0967212xxx <0967212xxx>->0033972480xxx in context public
>
>
>     <include>
>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>     <param name="username" value="0033972480xxx"/>
>     <param name="password" value="xxxxxxx"/>
>     <param name="proxy" value="sip.ovh.fr <http://sip.ovh.fr>"/>
>     <param name="expire-seconds" value="800"/>
>     <param name="register" value="true"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="extension" value="auto_to_user"/>
>     <param name="context" value="public"/>
>     </gateway>
>     </include>
>
>     I tried to edit my inbound dialplan manually, it works using
>     <condition field="${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
>
>

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