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@vocalseeds.com [mailto:danish.samad@vocalseeds.com] Sent: Tuesday, August 01, 2006 6:37 AM To: freeswitch-users@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@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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