[Freeswitch-users] Change invite header on inbound call

[Tanguy]

Hello, i found a another way to do this.

I tried like 
https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk. 
This is not exactly like a provider trunk because i have to change the 
dialplan if the remote server IP is changing, but for my immediate use, 
it will be enough.

Sorry for dump question.


<extension name="ringbacktone" continue="true">

<condition field="destination_number" expression="^(0\d{9})$">
<action application="ring_ready" />
<action application="bridge" data="sofia/external/$1 at asterisk_ip_adress"/>
</condition>
</extension>



On 14/06/2015 13:28, Tanguy wrote:
> Hello
>
> I would like to use a freeswitch server as gateway to share my inbound 
> trunks between several other servers (   asterisk and freeswitch ).  
> My trunks will be connected to the "gateway freeswitch server"
>
> On inbound call: some DID numbers should ring the production asterisk 
> server (  or the backup asterisk server if the production peer is not 
> registered )  Some others DID numbers will ring a another test 
> freeswitch server.
>
> I created several sip account on internal profile for each remote 
> server 1000( asterisk ) , 1001(asterisk-backup), 1002(freeswitch 
> testing server) and i tried to transfer a call like this ( i did not 
> implemented DID number filtering yet )
>
> *<condition>
> <action application="transfer" data="1000 XML default" />
> </condition>*
>
> When i call one of my public numbers the 1000 at default extension is 
> ringing but i don't like the SIP invite header
>
> *INVITE sip:1000 at 192.168.0.14:5070;transport=udp;user=phone SIP/2.0*
>
> I would like something like this to distinguish the destination number 
> on the remote servers.
>
> *INVITE sip:<public_destination_number>@ip_adresss:5060 SIP/2.0*
>
> As you can see, my headers may simulate the telco headers.
>
> I did not find how to change theses headers on the dialplan
>
> Best regards
>
>
>

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