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
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@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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