[Freeswitch-users] Disable reinvite

[Richard Klingler]

Hmm...

Doesn't help setting:

	<param name="inbound-bypass-media" value="false"/>

Seems to be som eother problem...

Internal phones are on network 10.0/16 and connect to FS via internal network...
But as soon RTP traffic is flowing, RTP traffic goes out via internet gateway
and back in into FS via its WAN IP. (Internet FW and FS WAN are one same /29 subnet).

So somehow FS thinks the internal phones have a public IP and not RFC1918.


On Mon, 2 Jul 2012 18:57:18 +0400, Yuriy Nasida wrote:
> You have to use bypass_media for it.
> 
> http://wiki.freeswitch.org/wiki/Bypass_Media 
> 
> 
> 
> 
>> Date: Mon, 2 Jul 2012 12:50:26 +0200
>> From: richard at klingler.net
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Disable reinvite
>> 
>> Hello
>> 
>> Just switched this weekend from asterisk to freeswitch on freebsd 9.0.
>> And so far I was able to implement basic dialplan for 
> inbound/outbound calls (o;
>> 
>> What I'm trying to find is a similar configuration option from 
> asterisk sip.conf
>> where individual user agents can have SIP reinvite disbaled with 
> "canreinvite=no".
>> 
>> Fomr the Freeswitch documentation it states that all RTP streams 
> are passed through
>> by default, but I still see SIP clients trying to send UDP streams 
> to my box directly.
>> 
>> Also setting "<param name="inbound-proxy-media" value="true"/>" in 
> sofia.conf.xml
>> doesn't solve this problem. I just want the UDP streams frmo 
> outside are only allowed
>> from my SIP trunk provider.
>> 
>> 
>> thanx in advance
>> richard
>> 
>> 
>> 
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> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
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