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

Papineni, Suneel SPapineni at enghouse.com
Mon Nov 21 13:28:54 MSK 2011


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<mailto: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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