Try changing these lines and put your ip instead, that worked for me. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
Diego On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote: > Ive Tried the following with no success: > > internal.xml > > <param name="ext-rtp-ip" value="75.101.142.208"/> > <param name="ext-sip-ip" value="75.101.142.208"/> > > Ive also tried just changing the vars.xml file. I also tried forwarding the > incoming rtp connections to my test pc. all with no audio success. I am > sure im doing something wrong I jsut cant find it. > > I found another suggestion on the wiki of creating a "double nat" that > listens on 5090. That didnt change anything either, here is that information: > > <settings> > <param name="debug" value="0"/> > <param name="sip-trace" value="no"/> > <param name="rfc2833-pt" value="101"/> > <param name="sip-port" value="5090"/> > <param name="dialplan" value="XML"/> > <param name="context" value="default"/> > <param name="dtmf-duration" value="100"/> > <param name="codec-prefs" value="$${outbound_codec_prefs}"/> > <param name="hold-music" value="$${hold_music}"/> > <param name="use-rtp-timer" value="true"/> > <param name="rtp-timer-name" value="soft"/> > <param name="manage-presence" value="false"/> > <param name="aggressive-nat-detection" value="true"/> > <param name="inbound-codec-negotiation" value="generous"/> > <param name="nonce-ttl" value="60"/> > <param name="auth-calls" value="false"/> > <param name="rtp-timeout-sec" value="1800"/> > <param name="rtp-ip" value="$${local_ip_v4}"/> > <param name="sip-ip" value="$${local_ip_v4}"/> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> > <param name="ext-sip-ip" value="$${external_sip_ip}"/> > <param name="rtp-timeout-sec" value="300"/> > <!--<param name="enable-3pcc" value="true"/>--> > <!-- TLS: disabled by default, set to "true" to enable --> > <param name="tls" value="false"/> > <!-- additional bind parameters for TLS --> > <param name="tls-bind-params" value="transport=tls"/> > <!-- Port to listen on for TLS requests. (5061 will be used if > unspecified)$ > <param name="tls-sip-port" value="5081"/> > <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for > $ > <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> > <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work > w$ > <param name="tls-version" value="tlsv1"/> > > </settings> > > Maybe someone can see what is going on here. > > Thanks, > Damon > > > -----Original Message----- > From: "Brian West" <[EMAIL PROTECTED]> > Sent: Friday, September 5, 2008 6:57pm > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 > > You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml > profile on ec2 duplicate them from the external profile. > > /b > > On Sep 5, 2008, at 8:47 PM, Damon Brown wrote: > >> Yes, I have all of the valid posts open on my security group >> -d > > Brian West > sip:[EMAIL PROTECTED] > > > > > > > > _______________________________________________ > 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 > > > > _______________________________________________ > 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 > _______________________________________________ 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