[Freeswitch-users] NO VOICE BETWEEN TWO REMOTE ENDPOINTS

[Brian West]

It could be many things... Firewall?  Nat being a nazi... what does  
your dialplan look like?  So many variables but from the info provided  
we need more to make a guess.

You'll need to provide a sip trace.  You can get that via starting  
FreeSWITCH like this TPORT_LOG=1 ./freeswitch

If you're on windows you can set the ENV variable to collect the same  
output

Then you can place that up on our paste bin located at http://pastebin:freeswitch@pastebin.freeswitch.org

Thanks,
/b
PS: service iptables stop



On Jul 7, 2008, at 4:03 PM, M.Emran wrote:

> 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:
>
> -
> <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  
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Brian West
sip:brian at freeswitch.org



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