[Freeswitch-users] How to interface with google talk

[danish.samad at vocalseeds.com]

Hi,

Mike, thanks for showing your concern. I was able to successfully bridge a
call between a google talk client and asterisk(SIP). Although there were
times when asterisk received and accepted the call while the jingle
session was dropped. I did not really delve into the issue, so I am not
really sure what was causing the issue.

There are certain things I noticed, that I would like to know more about.
Firstly, when calling from jinlge to sip,I can only map a single jabber
account with a SIP account. Is there any way I can bridge multiple calls
between Jingle and SIP endpoints, using the current configuration. If not,
are you guys planning to implement this feature.

Secondly, how can I make calls from sip endpoints to Jingle endpoints. Any
configuration tips will be helpful.

Regards,
Danish

> Danish, did you ever get a response to this message that resolved your
> issue?  We were all at our conference last week, so this may have been
> missed.
>
> Mike
>
> -----Original Message-----
> From: danish.samad at vocalseeds.com [mailto:danish.samad at vocalseeds.com]
> Sent: Tuesday, August 01, 2006 6:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] How to interface with google talk
>
> Hi,
>
> I am trying to interface with google talk using the following setup
>
> Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client
>
> Freeswitch will register with Jabberserver using a valid jabber account
> on
> the Jabber server. Using this account I should be able to login and add
> any valid gmail id in its userlist and perform standard operations
> (using
> the jingle protocol) such as chat, voice, etc. Eventually I intend to
> use
> Freeswitch as a gateway between SIP and google talk clients, but I first
> need to validate the above setup.
>
>  To test the setup out, I created an account on jabber.anywise.com. I
> was
> able to successfully log into the server using PSI and adding a valid
> gmail accounts in my roster.  Now when I specify the account in
> freeswitch.xml I am not able to register as I donot see the user coming
> online in my google talk user list. I just see the following XML
> messages
> being dumped on the console.
>
> freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h
> ttp://etherx.jabber.org/streams' xmlns='jabber:client'
> to='jabber.anywise.com' version='1.0'>]
>
> RECV[<?xml version='1.0'?><stream:stream
> xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7'
> xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>]
>
> RECV[<stream:features><register
> xmlns='http://jabber.org/features/iq-register'/><auth
> xmlns='http://jabber.org/features/iq-auth'/></stream:features>]
>
> For reference I am attaching related excerpts from the freeswitch.xml
> file
>
> <load module="mod_dingaling"/>
> ....
> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
>       <settings>
>         <param name="debug" value="1"/>
>         <param name="codec-prefs" value="PCMU"/>
>       </settings>
>       <interface>
>         <param name="name" value="jingle"/>
>         <param name="login" value="user at jabber.anywise.com/talk"/>
>         <param name="password" value="pwd"/>
>         <param name="dialplan" value="XML"/>
>         <param name="message" value="Jingle all the way"/>
>         <param name="rtp-ip" value="vostroip"/>
>         <param name="auto-login" value="true"/>
>         <param name="use-rtp-timer" value="true"/>
> 	<param name="exten" value="7777"/>
>         <param name="vad" value="both"/>
>       </interface>
> </configuration>
> ...
>
> <extension name="7777">
>  <condition field="destination_number" expression="^7777$">
>    <action application="bridge" data="playback
> /var/lib/asterisk/sounds/test.wav"/>
>  </condition>
> </extension>
>
> Please suggest what I need to do to achieve the required setup. Any help
> will be appreciated.
>
> Regards,
> Danish
>
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