Try turning of VAD and see if it will help. I am sure you should not be setting both proxy_media and bypass_media to true, as you should be transcoding if I remember correctly. http://wiki.freeswitch.org/wiki/Proxy_Media On Mon, Nov 21, 2011 at 12:28 PM, Papineni, Suneel <SPapineni@enghouse.com>wrote:
Hi,****
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I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in “client” profile.****
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What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on “client profile” and created a dial-plan for this number to ‘park’ the call first before trying to join to the conference.****
Then using eventSockets I am trying to join this call to conference and issued following command. (tried with “uuid_bridge” command as well)****
"api uuid_transfer [Unique-ID] conference:xyz@default inline"****
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Command is successful and also I can hear a sound that someone joined in the conference, but I didn’t hear any voice at either side. I couldn’t see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically.****
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I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue.****
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Could someone please let me know if I am missing anything or need to configure in a different way for conferencing.****
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Thanks & Regards****
Suneel****
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*Client.xml*
<profile type="client">****
<param name="name" value="gmail.com"/>****
<param name="login" value="user@gmail.com/gtalk"/>****
<param name="password" value="password"/>****
<param name="dialplan" value="XML"/>****
<param name="context" value="public"/>****
<param name="message" value="Jingle all the way"/>****
<param name="rtp-ip" value="$${bind_server_ip}"/>****
<param name="auto-login" value="true"/>****
<param name="sasl" value="plain"/>****
<param name="server" value="talk.google.com"/>****
<param name="tls" value="true"/>****
<param name="use-rtp-timer" value="true"/>****
<param name="exten" value="123456"/>****
<param name="vad" value="both"/>****
</profile>****
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*Dial-plan..*
<include>****
<extension name="GTalk_DP">****
<condition field="caller_id_number" expression="^([^@]+)" break="never">****
<action application="set" data="effective_caller_id_number=$1" />****
</condition>****
<condition field="destination_number" expression="^(.*)$">****
<action application="set" data="proxy_media=true"/>****
<action application="set" data="bypass_media=true"/>****
<action application="park" />****
</condition>****
</extension>****
</include>****
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