Hello, I've created a doublenat profile and the users are normally registered. When I try to cal PSTN the phone answer but the parts cannot hear each other. The firewall is disabled. Any clues ? Please check below thanks Duan freeswitch@maui> 2008-08-25 10:23:26 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/doublenat/1000@voipclic.com:5090 [b67119fe-c344-4a4f-9496-d9e1f31174bf] 2008-08-25 10:23:26 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Cliconnect->011555133349905@default 2008-08-25 10:23:26 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/011555133349905 [9c023b8a-204a-4128-b3f3-76788f9246bf] 2008-08-25 10:23:28 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/external/011555133349905! 2008-08-25 10:23:28 [NOTICE] sofia_glue.c:2010 sofia_glue_tech_media() Pre-Answer sofia/external/011555133349905! 2008-08-25 10:23:28 [INFO] mod_sofia.c:1072 sofia_receive_message() Asked to send early media by sofia/doublenat/1000@voipclic.com:5090 2008-08-25 10:23:28 [INFO] mod_sofia.c:1113 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1219656570 1219656571 IN IP4 24.67.78.200 s=FreeSWITCH c=IN IP4 24.67.78.200 t=0 0 a=sendrecv m=audio 28438 RTP/AVP 0 101 13 a=rtpmap:0 G711U/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2008-08-25 10:23:28 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/doublenat/1000@voipclic.com:5090! 2008-08-25 10:23:28 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/doublenat/1000@voipclic.com:5090! 2008-08-25 10:23:37 [NOTICE] sofia.c:2439 sofia_handle_sip_i_state() Channel [sofia/external/011555133349905] has been answered 2008-08-25 10:23:37 [NOTICE] sofia.c:2443 sofia_handle_sip_i_state() Channel [sofia/doublenat/1000@voipclic.com:5090] has been answered <profile name="doublenat"> <gateways> <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> </gateways> <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="$${outbound_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="apply-nat-acl" value="rfc1918"/> <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="force-register-domain" value="$${domain}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>