[Freeswitch-users] NO VOICE BETWEEN TWO ENDPOINTS UNDER NAT

[M.Emran]

Hi,

I have configured FS like this.but voice is not transmitting between two
endpoints.Can you help me where i did wrong ? Here is my configuration:

 - <#11afef267e7502b2_>
<profile name="internal" domain="$${domain}">
  <aliases>
    <alias name="$${domain}"/>
    <alias name="default"/>
  </aliases>
  <gateways>
    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  </gateways>
   <domains>
  </domains>

  <settings>
    <param name="debug" value="1"/>
    <param name="sip-trace" value="no"/>
    <param name="context" value="public"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- port to bind to for sip traffic -->
    <param name="sip-port" value="60"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${global_codec_prefs}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="aggressive-nat-detection" value="true"/>
    <!-- ip address to use for rtp -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <!-- ip address to bind to -->
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="rfc1918"/>
    <param name="record-template"
value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <param name="manage-presence" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="tls" value="false"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="5061"/>
    <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
    <param name="tls-version" value="tlsv1"/>

   <param name="inbound-no-media" value="true"/>


    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="true"/>
    <param name="auth-all-packets" value="false"/>
   <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>

  </settings>
</profile>





-- 
Regards
----------
M Emran
Inspiration Software Ltd
Web: www.inspiresoftbd.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080708/730cdc8c/attachment-0002.html