[Freeswitch-dev] NAT issue with SBC
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/${destination_number}@192.168.1.123"/> </condition> </condition> </extension> Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK. if any one can help, thank all alot. I am getting this stage to long and cant find any solution here. Thanks, ThanhTruong
Hi Thanh, my 2c to help you going through this: - If asterisk keeps re-senging the 200 OK after an ACK was sent to it, you should check whether the ACK was actually delivered, and if delivered whether asterisk logged any error/warning about it. In general, you should provide the SIP traces to get some help. - You have a typo in your public.xml: 'asteirsk' instead of 'asterisk'. You should check that. (BTW I'm not sure this is a question for -dev, perhaps -users?) Regards, Giacomo On 7 September 2013 19:30, ThanhTruong <ngoc217thanh@yahoo.com> wrote:
Hi all,
I have configure freeswitch as SBC server with one asterisk server.
I am getting nat issue when make call from FS to FS.
my case as:
client --- FS ----- Asterisk server ----- FS -----client.
my configure in public.xml like:
<extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" 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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participants (2)
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Giacomo Vacca
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ThanhTruong