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
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@klingler.net To: freeswitch-users@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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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@klingler.net To: freeswitch-users@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Why do you set false ? Do you want to send RTP media without FS, right ?So you can simply set <action application="set" data="bypass_media=true"/> in dialplan.
Date: Fri, 6 Jul 2012 13:37:12 +0200 From: richard@klingler.net To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Disable reinvite
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@klingler.net To: freeswitch-users@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
-
Richard Klingler
-
Yuriy Nasida