[Freeswitch-users] sip_to_user and destination number

[Chris Tunbridge]

Tanguy,

setting the auto_to_user should set the destination number field to the
correct one according to the wiki

"Note: *extension* parameter influence the contents of channel variable
*Caller-Destination-Number* and *destination_number*. If it is blank,
*Caller-Destination-Number* will always be set to gateway's username. If it
has a value, *Caller-Destination-Number* will always be set to this value.
If it has value *auto_to_user*, *Caller-Destination-Number* will be
populated with value *${sip_to_user}* which means the real dialed number in
case of an inbound call."

from: https://wiki.freeswitch.org/wiki/Sofia.conf.xml

where is this config that you listed currently located? (full file path
please)

On Fri, May 1, 2015 at 2:03 PM, Tanguy <phenix at vfemail.net> wrote:

>  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> 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
>>
>>
>>
>
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