Re: [Freeswitch-users] How to interface with google talk
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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From: danish.samad@vocalseeds.com [mailto:danish.samad@vocalseeds.com]
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.
I would like to see some traces of what is going on in freeswitch's debug output when this happens.
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.
The main issue here is a limitation in the jingle protocol, and the gtalk client. If using mod_dingaling on both sides with freeswitch, we have an extension to the protocol that lets you pass the extension. If using gtalk, you would need a way to gather digits such as in a javascript and launch that call to an extension. The dingaling channel will take text messages in the call beginning with + to be dtmf.
Secondly, how can I make calls from sip endpoints to Jingle endpoints. Any configuration tips will be helpful.
You need to do it in some way that you know what destination and profile will be used to call out, as for dingaling you need to specify the sending profile. You could just hardcode it to a specific profile. We use regex to parse a string like profilename!username!domain.com and then send that as sip, parse the 3 parts using regex, and send it to the bridge command for dingaling. Mike.
participants (2)
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danish.samad＠vocalseeds.com
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Michael Jerris