I would start by getting the server on a public ip, we actually filter out the private ip's when we handle candidates from clients, so it will probably NOT work at all that way. Mike On Apr 4, 2008, at 9:44 AM, Ali Jawad wrote:
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@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@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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