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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