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