[Freeswitch-users] FS and Gtalk integration

[jesse zhao]

Any expert on this issue?

-jesse

On Wed, Aug 18, 2010 at 5:05 PM, jesse zhao <chat2jesse at gmail.com> wrote:
> hi, help:
>
>  I searched online pages and FS wiki , basically the doc about Gtalk
> and FS is pretty lousy. I still couldn't get
> my gtalk and FS work together.
>
>   two major issues:
>
>  1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established,
> but no audio.
>  2) how could I call SIP from gtalk ?  when I request chat to sip,
> gchat in gmail requests an invitation.
>     suppose the voice request is : conf+1000 at 172.18.115.73
>
>  here are my config files:
>  jingle_profiles/client.xml
>
>  <profile type="client">
>    <param name="name" value="gmail.com"/>
>    <param name="login" value="XYZ at gmail.com/talk"/>
>    <param name="password" value="XXXXX"/>
>    <param name="dialplan" value="XML"/>
>    <param name="context" value="default"/>
>    <param name="message" value="Jingle all the way"/>
>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="auto-login" value="true"/>
>    <!-- SASL "plain" or "md5" -->
>    <param name="sasl" value="md5"/>
>    <!-- if the server where the jabber is hosted is not the same as
> the one in the jid -->
>    <param name="server" value="talk.google.com"/>
>    <!-- Enable TLS or not -->
>    <param name="tls" value="true"/>
>    <!-- disable to trade async for more calls -->
>    <param name="use-rtp-timer" value="true"/>
>    <!-- default extension (if one cannot be determined) -->
>    <param name="exten" value="jingle2sip"/>
>    <!-- VAD choose one -->
>    <!-- <param name="vad" value="in"/> -->
>    <!-- <param name="vad" value="out"/> -->
>    <param name="vad" value="both"/>
>    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>    <!--<param name="candidate-acl" value="wan.auto"/>-->
>    <param name="local-network-acl" value="localnet.auto"/>
>  </profile>
> </include>
>
>
>  dialplan/default.xml
>
>    <extension name="jingle2sip">
>      <condition field="source" expression="mod_dingaling"/>
>      <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$">
>       <action application="bridge" data="sofia/default/$1%172.18.115.73"/>
>      </condition>
>    </extension>
>
>    <extension name="sip2jingle">
>          <condition field="source" expression="mod_sofia"/>
>          <condition field="destination_number"
> expression="^gtalk=([a-zA-z0-9.-]+)$">
>            <action application="set" data="no_media=false"/>
>            <action application="set"
> data="effective_caller_id_number=myuser at gmail.com"/>
>            <action application="bridge"
> data="dingaling/gmail.com/$1 at gmail.com"/>
>          </condition>
>
> I tried suggestion from
> http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456.html,
> above two issues still not work.
>
>  I heard id: intralanman is good at this, if you saw the post, please
> share your advice.
>
> -jesse
>