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="[EMAIL PROTECTED]/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 [EMAIL PROTECTED]
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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