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Hello, i found a another way to do this.<br>
<br>
I tried like
<a class="moz-txt-link-freetext" href="https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk">https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk</a>.
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.<br>
<br>
Sorry for dump question.<br>
<br>
<br>
<extension name="ringbacktone" continue="true"><br>
<br>
<condition field="destination_number"
expression="^(0\d{9})$"><br>
<action application="ring_ready" /><br>
<action application="bridge"
data="sofia/external/$1@asterisk_ip_adress"/><br>
</condition><br>
</extension><br>
<br>
<br>
<br>
On 14/06/2015 13:28, Tanguy wrote:
<blockquote cite="mid:557D657B.6060007@vfemail.net" type="cite">
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charset=ISO-8859-1">
Hello<br>
<br>
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"<br>
<br>
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.<br>
<br>
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 )<br>
<br>
<b><tt><condition> <br>
<action application="transfer" data="1000 XML default"
/><br>
</condition></tt></b><br>
<br>
When i call one of my public numbers the 1000@default extension is
ringing but i don't like the SIP invite header<br>
<br>
<b><tt>INVITE <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:sip:1000@192.168.0.14:5070;transport=udp;user=phone">sip:1000@192.168.0.14:5070;transport=udp;user=phone</a>
SIP/2.0</tt></b><br>
<br>
I would like something like this to distinguish the destination
number on the remote servers.<br>
<br>
<tt><b>INVITE
sip:<public_destination_number>@ip_adresss:5060 SIP/2.0</b></tt><br>
<br>
As you can see, my headers may simulate the telco headers.<br>
<br>
I did not find how to change theses headers on the dialplan<br>
<br>
Best regards<br>
<br>
<br>
<br>
</blockquote>
<br>
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