[Freeswitch-users] My FS + Dingaling Experience With Qs

[Ali Jawad]

Hi All

 

I am trying to test the jingle features of FS. I have compiled and setup
FS on debian all went well and I made a call using softphones from 1000
to 1001. After that I read

http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working

 

I did the changes and rebuilt

 

I edited my config files my  ./conf/jingle_profiles/client.xml looks as
follows

 

<include>

  <!-- Client Profile (Original mode) -->

  <x-profile type="client">

    <param name="name" value="gmail.com"/>

    <param name="login" value="thisisme at gmail.com/talk"/>

    <param name="password" value="thisIsMyPwd"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="message" value="Jingle all the way"/>

    <param name="rtp-ip" value="auto"/>

    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip"/> -->

    <param name="auto-login" value="true"/>

    <param name="auto-reply" value="Press *Call* to join my
conference"/>

    <param name="exten" value="1000"/>

    <param name="ext-rtp-ip" value="stun:stun.xten.com"/>

    <!-- SASL "plain" or "md5" -->

    <param name="sasl" value="plain"/>

    <!-- if the server where the jabber is hosted is not the same as the
one in the jid -->

    <!--<param name="server" value="alternate.server.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="888"/>

    <!-- 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"/>-->

  </x-profile>

</include>

 

As you can see above I used    <param name="exten" value="1000"/> after
that I logged in with my SIP softphone "Eyebeam" and tried to make a
call to gmail account that did not work..but I was able to make a
conference call to conf+888 at conference.freeswitch.org

 

Please note that my clients and server are both running in the same
private LAN, I can put the server on a public IP if necessary. 

 

So what must I do/change/add/check to able to make calls to my gmail
accounts (or from my gmail account to my sip phone) as mentioned in the
FAQ

 


Q: What? Did you say it can talk to GoogleTalk?


Yes in March of 2006 I developed my own XMPP telephony signaling library
that is capable of communicating with Google's GoogleTalk. With a single
Jabber account you can receive endless simultaneous calls from
GoogleTalk clients and gateway those calls to IVR or another voice
protocol like SIP or H.323. When FreeSWITCH is on both ends of the call
you can bypass NAT and send extended data such as Caller ID and DNIS. 

 

 

 

 

 

 

 

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