Hi Anthony

 

Thx for your help so far, I am sure it work but I have some misconfig J.

I have enabled debug in dingaling.conf.xml and I can see the presence of my gtalk friends when they sign in and out of the console I have also added this statement to my dialplan/default.xml 

 

<extension name="sip2jingle">

  <condition field="source" expression="mod_sofia"/>

  <condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$">

    <action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/>

  </condition>

</extension>

 

I had previously missed this one, I found it at http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/ since I could not find any jingle bridge examples in the wiki.

 

However this did not add any info to the debug console, and the failure is still the same

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Tuesday, April 08, 2008 4:25 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] About Dingaling

 

also enable the debug option in dingaling.conf.xml and set the default level to debug in console.conf.xml to see the xmpp traffic on the console.

I can attest it works, you just need to configure it properly.

On Tue, Apr 8, 2008 at 8:10 AM, Ali Jawad <ali.jawad@splendor.net> wrote:

Thx anthony

 

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

 

someaccount@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Tuesday, April 08, 2008 4:00 PM
To: freeswitch-users@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@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@gmail.com" claiming to be MailScanner has detected a possible fraud attempt from "thisisme@gmail.com" claiming to be 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

 

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@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

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