[Freeswitch-users] Regarding Freeswicth Nat

johnthan123 johnthan123 at gmail.com
Wed Jul 3 16:42:40 MSD 2013

Hi All Finally i Figured Out the NAT issue

My Setup is Like this

DID(IP Authentication)---->Public IP----------> (Pfseance)(1:1)NAT
--------> (Private IP) FreeSwicth------->(Private IP)Asterisk Cluster.

        i am not sure why its work for Me, Really :), Because with this
same configuration its didn't work for me until yesterday.

Here the Configurations is :

First thing  we need the set the variable for   vars.xml

XX.XX.XX.XX = your Public IP

<X-PRE-PROCESS cmd="set" data="external_sip_ip=XX.XX.XX.XX"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=XX.XX.XX.XX"/>

Add the IP address of Your Provider to acl:

YY.YY.YY.YY = Your Provider IP

<node type="allow" cidr="YY.YY.YY.YY/32"/>

Add the Dialplan Entry in Public :

 <extension name="public_did">
    <condition field="destination_number" expression="2064248436">
    <action application="set" data="domain_name=$${domain}"/>
    <action application="bridge" data="sofia/gateway/VICI22/2064248436"/>

This is it  i am Done, my Asterisk Cluster is reciving calls with two way

May be this will Help For Some People to solve the NAT issue.


On Mon, Jul 1, 2013 at 5:39 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Mon, Jul 1, 2013 at 1:37 PM, johnthan123 <johnthan123 at gmail.com> wrote:
>> Hi All,
>>  I am really happy for the all answers, But the thing is No one giving me
>> the assure for the NAT,
>>              i configured the NAT before and it was working, But with the
>> same configuration and same networking i couldn't able to make it now, i
>> have similor qustion with asterisk why asterisk works good behind the NAT
>> why not Freeswitch,  as a Freeswitch Lover i want to be overcome this issue
>>         still i am looking for some inputs for successful NAT in
>> Freeswitch , if possible some one give for NAT.
> The short answer is this: something somewhere changed. If it was working
> with a previous version of FS and isn't working now then I would recommend
> looking for the commit where it broke. Install an older FS version where
> you are certain that it was working, and then go from there. Using git
> bisect can probably help cut down the time.
> I would also recommend getting SIP and RTP captures of working vs.
> non-working calls and compare them using Wireshark. See if you can figure
> out what is happening and then report back. Hopefully the gang here will
> help you figure out why it is happening and you can fix it.
> -MC
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