[Freeswitch-users] "To field" in sip header

[thomas peterseil]

hi,
thanks a lot, that solved my problem!
but i am still wondering if it´s possible to read and modify sip header
fields with FS at inbound calls.

best,
thomas
Am 29.04.2016 8:54 nachm. schrieb "Tanguy" <phenix at vfemail.net>:

> Hello
>
> I think you may add this line in your gateway configuration.
>
> <param name="extension" value="auto_to_user"/>
>
> Then you will be able to use a standard dialplan with destination_number
>
>
>
> On 29/04/2016 14:47, thomas peterseil wrote:
>
> hello,
> i have a sip provider with multiple DID´s and they send me all calls with
> the same invite, only the To field in the SIP header is different. how can
> i read the To field in the SIP header to be able to make a call routing
> based on the To field? i tried the following rule but it doesnt work:
>
> ---------------------
>
> <include>
>
> <extension
> name="public_test1">
>
>         <condition
> regex="all">
>
>         <regex  field="destination_number"
> expression="^provider1$"/>
>         <regex  field="${sip_h_To}"
> expression="43717072377"/>
>
>
>                 <action application="set"
> data="effective_caller_id_number=7021"/>
>                 <action application="set"
> data="effective_caller_id_name=thomas"/>
>                 <action
> application="answer"/>
>
>                 <action application="transfer" data="6011 XML inbound1
> "/>
>
> </condition>
>
> </extension>
>
> ----------------
>
> the To field in the SIP header looks like this:
>
> To: <sip:43717072377 at 193.43.148.37>
>
> thanks a lot for help!
>
> best regards,
> thomas
>
>
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