[Freeswitch-users] About Dingaling

Tue Apr 8 06:10:17 PDT 2008

Thx anthony 

I tried that and then I opened my SIP phone and I dialed

someaccount at gmail.com and it said Call failed : Request Timed out

I changed to <client></client> with the same results.

I must add that I did not change anything in server.xml which has
default values and is set to component mode ..should I change it too ?

thx

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Tuesday, April 08, 2008 4:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] About Dingaling

try taking the x- out of the tags
change <x-client> ... </x-client> to <client> ... </client>

The x- in the tag name is a way to comment it out.

On Tue, Apr 8, 2008 at 5:46 AM, Ali Jawad <ali.jawad at splendor.net>
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="MailScanner has detected a possible fraud
attempt from "thisisme at gmail.com" claiming to be thisisme at gmail.com/talk
<http://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 at conference.freeswitch.org

The server is running on a PUBLIC IP

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.

And where can  I debug this issue, if I call 9999 I can see the call
setup and successful call in my putty shell, however when I call
myaccount at gmail.com nothing happens.

Thx

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. 

With Regards

Ali Jawad

System Administrator

Splendor Telecom (www.splendor.net)

Beirut, Lebanon

Phone: +961 1 373725

Fax: + 961 1 375554

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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
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