[Freeswitch-users] No Voice

[Cliconnect]

I was having a similar problem just yesterday and it was a firewall problem.

 
Check this page and make sure to forward and open the proper ports.
 
 <http://wiki.freeswitch.org/wiki/Getting_Started_Guide>
http://wiki.freeswitch.org/wiki/Getting_Started_Guide 
 
 
Duan

 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of M.Emran
Sent: Tuesday, August 26, 2008 7:53 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] No Voice


i have installed FS in public.And configured profile as follows:

<profile name="doublenat">
    <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="6000"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <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="true"/>
    <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>

my dialplan is like this :
<extension name="echo test">
        <condition field="destination_number" expression="999">
         <action application="answer" />
         <action application="echo"/>
        </condition>
      </extension>
      
 
      <extension name="curl_test">
        <condition field="destination_number" expression="(.*)">
        <action application="set" data="proxy_media=true"/>
        <action application="bridge" data="sofia/internal/$1%$${domain}" />
        <action application="hangup" />
        </condition>
      </extension>


I have called from my pc to another pc with X-Lite. I can not hear voice
from any end. i have dialed 999 for echo test.that also not working.

any help will be appreciatable.


-- 
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/20080826/070c2c4c/attachment-0002.html