[Freeswitch-users] How to interface with google talk

[Blaine Cook]

Everything you're doing looks correct; this is the extension that worked for
me:

<extension name="999">
   <condition><action application="bridge" data="iax/user at ast.eri.sk/5551212
"/></condition>
</extension>

my guess is that the condition in your extension config is what's causing
the failure.

b.

On 8/1/06, danish.samad at vocalseeds.com <danish.samad at vocalseeds.com> wrote:
>
> 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
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20060803/b1f615e1/attachment-0003.html