[Freeswitch-users] sip_to_user and destination number

[Stanislav Sinyagin]

Remove the extension parameter and see if it helps.
On May 1, 2015 6:11 PM, "Tanguy" <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
> SIP/2.0.
> Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr.
> Contact: <sip:10.7.1.60:5060>.
> Content-Type: application/sdp.
> CSeq: 403749831 INVITE.
> From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
> <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>
> <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,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am
> unable to do the same under freeswitch.
>
> My trunk configuration seems correct, as you can see i used  auto_to_user,
> but the destination number remains 0033972480xxx when i call 0557590xxx.
>
> 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing
> 0967212xxx <0967212xxx>->0033972480xxx in context public
>
>
> <include>
>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>       <param name="username" value="0033972480xxx"/>
>       <param name="password" value="xxxxxxx"/>
>       <param name="proxy" value="sip.ovh.fr"/>
>       <param name="expire-seconds" value="800"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>       <param name="extension" value="auto_to_user"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> I tried to edit my inbound dialplan manually, it works using  <condition
> field="${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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150501/0d87751f/attachment.html