[Freeswitch-users] Joining a call from Gtalk to a conference on FreeSwitch

[Kerem Erciyes]

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 at enghouse.com>wrote:

>  Hi,****
>
> ** **
>
> 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.****
>
> ** **
>
> 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 at default inline"****
>
> ** **
>
> 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.****
>
> ** **
>
> 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.****
>
> ** **
>
> Could someone please let me know if I am missing anything or need to
> configure in a different way for conferencing.****
>
> ** **
>
> Thanks & Regards****
>
> Suneel****
>
> * *
>
> *Client.xml*
>
> <profile type="client">****
>
>     <param name="name" value="gmail.com"/>****
>
>     <param name="login" value="user at 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>****
>
> ** **
>
> *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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>
> 
> 
>
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-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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