[Freeswitch-users] No sound

[Cliconnect]

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 at maui> 2008-08-25 10:23:26 [NOTICE] switch_channel.c:534
switch_channel_set_name() New Channel sofia/doublenat/1000 at 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 at 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 at 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 at voipclic.com:5090!
2008-08-25 10:23:28 [NOTICE] mod_sofia.c:1116 sofia_receive_message()
Pre-Answer sofia/doublenat/1000 at 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 at 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>