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@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@default extension is ringing but i don't like the SIP invite header

INVITE sip:1000@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