[Freeswitch-users] About Dingaling

Anthony Minessale anthony.minessale at gmail.com
Tue Apr 8 06:24:58 PDT 2008


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 at splendor.net> wrote:

>  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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> --
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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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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