have you looked at a Sip trace from both sides?  Is the ACK getting to asterisk?  Is it not accepting it for some reason?

On Sep 9, 2013, at 5:31 AM, ThanhTruong <ngoc217thanh@yahoo.com> wrote:


Hi all,

I have configure freeswitch as SBC server with one asterisk server.

I am getting nat issue when make call from FS to FS.

my case as:

client --- FS ----- Asterisk server ----- FS -----client.

my configure in public.xml like:

<extension name="asteirsk">
        <condition field="destination_number" expression="(\d+)$" require-nested="true">
            <condition field="network_addr" expression="^192\.168\.1\.123$"  break="never">
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
                <action application="set" data="bypass_media=true"/>
                <action application="bridge" data="user/${destination_number}@${domain_name}"/>
                <anti-action application="set" data="hangup_after_bridge=true"/>
                <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
                <anti-action application="set" data="bypass_media=true"/>
                <anti-action application="bridge" data="sofia/internal/${destination_number}@192.168.1.123"/>
            </condition>
        </condition>
    </extension>

Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK.

if any one can help, thank all alot. I am getting this stage to long and cant find any solution here.

Thanks, 
 
ThanhTruong




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