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:
<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>