[Freeswitch-users] Dingaling In Component Mode "How To CallJingleTo SIP"

Ali Jawad ali.jawad at splendor.net
Mon Apr 14 06:45:12 PDT 2008


I did that and I do have a user online called ext+1000 and I do have a
route setup as follows in my dialplan "This is the first route"


       <extension name="default">

                      <condition field="fdnis" expression=""/>

                      <condition field="destination_number"

                       <action application="bridge"




And I do have a gateway called c5gateway so as per this extension all
incoming calls should pass here right ?

This works in gtalk, did I miss something here ?


Thanks Anthony for your help. I really would like to get this
working..my client is jabbin by the way.


From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Monday, April 14, 2008 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dingaling In Component Mode "How To
CallJingleTo SIP"


The easiest way to do it in component mode is to use the special ext
name space which has a permanent active status.
Add ext+1234 at mydomain.com <mailto:ext%2B1234 at mydomain.com>  to your
buddy list.  if this leads to FreeSWITCH component it will automatically
add the buddy and mark
it active at all times.  Then when you call this id it will hit the dial
plan with the exact string ext+1234 at mydomain.com
<mailto:ext%2B1234 at mydomain.com>  as the destination number which you
can parse with regex and route to sip.

On Mon, Apr 14, 2008 at 1:30 AM, Ali Jawad <ali.jawad at splendor.net>

Dear All


This might be a duplicate post, I am sorry if it is. Normally I get
direct feedback from the list. But this time I did not get any reply. So
I was afraid it might not have been delivered.


First I would like to state that dingaling is working in client mode
with gtalk perfectly. I can call from gtalk to my sip extension "1000".
I have setup component mode now I am using FS with Openfire, open does
show the following:


1           fs.jabber2.mydomain.net             LibDingaLing


However stats on the jabber server are Session Statistics:
Packets Received/Sent: 1/1



So the component is REGISTERED, so far so good my problem now is how to
initiate a call from XMPP to a SIP extension. Using client mode I would
do the following:


1.	Setup GTALK account and assign extension to it 
2.	Login with another gtalk account and dial the first gtalk
3.	The assigned extension rings 


I don't know how to do the same using component mode. Adding
sip+1000 at mydomain to the jabber client "Jabbin" does not popup an "add
contact" on XTEN. 


Anything I missed here ?


Thanks for any hints or suggestions 



The Freeswitch server logs are below:


2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND:


<stream:stream xmlns:stream="http://etherx.jabber.org/streams"
xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream>


2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV:


<stream:stream xmlns:stream="http://etherx.jabber.org/streams"
xmlns="jabber:component:accept" from="fs.jabber2.MyDomain"


2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND:




2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV:




2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV:


<iq type="get" id="970-504" from="component.jabber2.MyDomain"

  <query xmlns="http://jabber.org/protocol/disco#info"></query>



2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND:


<iq from="fs" to="component.jabber2.MyDomain" id="970-504"

  <query xmlns="http://jabber.org/protocol/disco#info">

    <identity category="gateway" name="LibDingaLing"></identity>

    <feature var="http://jabber.org/protocol/disco"></feature>

    <feature var="http://jabber.org/protocol/disco#info"></feature>

    <feature var="jabber:iq:version"></feature>

    <feature var="vcard-temp"></feature>


    <feature var="jabber:iq:last"></feature>





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Anthony Minessale II

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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
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