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monemran at gmail.com Guest
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Posted: Tue Aug 26, 2008 9:55 am Post subject: [Freeswitch-users] No Voice |
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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 |
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cliconnect at cliconne... Guest
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Posted: Tue Aug 26, 2008 10:17 am Post subject: [Freeswitch-users] No Voice |
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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
Duan
Quote: |
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of M.Emran
Sent: Tuesday, August 26, 2008 7:53 AM
To: freeswitch-users@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
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