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

Mon Aug 11 21:28:42 MSD 2014

If you are still having an issue, please file a jira and attach debug information there.

On Apr 14, 2008, at 10:17 AM, Ali Jawad <ali.jawad at splendor.net> wrote:

> Hey Anthony
>  
> Thx for ur help again. I did use the extension you provided and I did set console to debug mode. Adding to that I do have sip and dingaling in debug mode and sip trace set to on in the outbound.xml so if a call is triggered to SIP I can see it immediately. However I doubt that the jingle call is transformed to SIP. The trace for a call is
>  
> http://pastebin.com/m5ea1b28b
>  
> I saw a lot of
>  
> 2008-04-14 13:06:11 [DEBUG] libdingaling.c:448 parse_session_code() Duplicate Pref!
>  
> And at the end there was
>  
> ·  2008-04-14 13:06:21 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill DingaLing/new [KILL]
> ·  2008-04-14 13:06:21 [DEBUG] libdingaling.c:284 ldl_session_destroy() Destroyed Session 1747058827
>  
>  
> But not SIP traces
>  
> Can you please take a look at it Anthony ?
>  
> Thx
>  
> 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 5:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Dingaling In Component Mode "How ToCallJingleTo SIP"
>  
> Try this:
> 
> If you issue "console loglevel debug" first on the cli you should see what the call is doing.
> 
>  
> 
>        <extension name="default">
> 
>                       <condition field="destination_number" expression="^ext\+1000"/>
> 
>                        <action application="bridge" data="sofia/gateway/c5gateway/009613041705"/>
> 
>                       </condition>
> 
>        </extension>
> 
>  
> 
> On Mon, Apr 14, 2008 at 8:45 AM, Ali Jawad <ali.jawad at splendor.net> wrote:
>  
> 
> 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" expression="">
> 
>                        <action application="bridge" data="sofia/gateway/c5gateway/009613041705"/>
> 
>                       </condition>
> 
>        </extension>
> 
>  
> 
> 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 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 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> wrote:
> 
> 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      gateway
> 
>  
> 
> 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:
> 
>  
> 
> Setup GTALK account and assign extension to it
> Login with another gtalk account and dial the first gtalk account
> 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" id="69fcaa7b"></stream:stream>
> 
>  
> 
> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND:
> 
> -------------------------------------------------------------------------------
> 
> <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake>
> 
>  
> 
> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV:
> 
> -------------------------------------------------------------------------------
> 
> <handshake></handshake>
> 
>  
> 
> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV:
> 
> -------------------------------------------------------------------------------
> 
> <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain">
> 
>   <query xmlns="http://jabber.org/protocol/disco#info"></query>
> 
> </iq>
> 
>  
> 
> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND:
> 
> -------------------------------------------------------------------------------
> 
> <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result">
> 
>   <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="http://www.google.com/xmpp/protocol/voice/v1"></feature>
> 
>     <feature var="jabber:iq:last"></feature>
> 
>   </query>
> 
> </iq>
> 
>  
> 
>  
> 
> 
> _______________________________________________
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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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400 
> -- 
> This message has been scanned for viruses and 
> dangerous content by MailScanner, and is 
> believed to be clean.
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