Dingaling In Component Mode "How To Call JingleTo SIP"
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: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@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>
Adding to the below, obviously I was doing something wrong because I did not use the xmpp subdomain reserved by the component created between my XMPP server and the freeswitch server. Now if I use my jingle client to add a user such as 1000@FScomponent.myXMPPdomain.net in the jingle client I get [11:34:03 AM] *** 1000@ FScomponent.myXMPPdomain.net is Online [Authenticated. Come to ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ] As for the relevant FS dingaling log I get http://pastebin.com/m36b3b9f1 In server.xml I have set the following <param name="exten" value="_auto_"/> Now my question is the following: * If I do call 1000@... Shouldn't extension 1000 on my sip device ring automatically or should I edit my settings to do that ? * I am seeing a lot of component.mydomain.net "did not add that myself" should the component.mydomain.net point the xmpp or FS server Any clues or help is appreciated. ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Jawad Sent: Monday, April 14, 2008 9:30 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Dingaling In Component Mode "How To CallJingleTo SIP" 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: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@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> -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.
Adding to the below there is one thing I really don't get If I register a gmail account using type=client I can call that account from a gtalk account and as per dialplan my cellphone rings If I register a typical xmpp account using type=client I can call that account from a jingle account I see the jingle call incoming to the FS server but that's it. It does not switch to SIP. If I use component mode and I add a contact to my xmpp client I can see the added contact online and I can call it but it does not transform to SIP. Does dingaling only work for GTALK or should it work for an ordinary XMPP server too ? Thx ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Jawad Sent: Monday, April 14, 2008 11:49 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] UPDATED -- Dingaling In Component Mode "How To CallJingleTo SIP" Adding to the below, obviously I was doing something wrong because I did not use the xmpp subdomain reserved by the component created between my XMPP server and the freeswitch server. Now if I use my jingle client to add a user such as 1000@FScomponent.myXMPPdomain.net in the jingle client I get [11:34:03 AM] *** 1000@ FScomponent.myXMPPdomain.net is Online [Authenticated. Come to ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ] As for the relevant FS dingaling log I get http://pastebin.com/m36b3b9f1 In server.xml I have set the following <param name="exten" value="_auto_"/> Now my question is the following: * If I do call 1000@... Shouldn't extension 1000 on my sip device ring automatically or should I edit my settings to do that ? * I am seeing a lot of component.mydomain.net "did not add that myself" should the component.mydomain.net point the xmpp or FS server Any clues or help is appreciated. ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Jawad Sent: Monday, April 14, 2008 9:30 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Dingaling In Component Mode "How To CallJingleTo SIP" 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: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@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> -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.
hello i am doing exactly thesame thing. i configured dingaling component mode on the server.xml , and the connexion between openfire and freeswitch works fine, but i can't make calls from openfire client to freeswitch client (i am using jitsi) if you have any solution? thank you
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@mydomain.com <ext%2B1234@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@mydomain.com <ext%2B1234@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@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:
1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings
I don't know how to do the same using component mode. Adding sip+1000@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>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, April 14, 2008 4:33 PM To: freeswitch-users@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@mydomain.com <mailto:ext%2B1234@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@mydomain.com <mailto:ext%2B1234@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@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: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@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> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.
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@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@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, April 14, 2008 4:33 PM *To:* freeswitch-users@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@mydomain.com <ext%2B1234@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@mydomain.com <ext%2B1234@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@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:
1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings
I don't know how to do the same using component mode. Adding sip+1000@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>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- This message has been scanned for viruses and dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is believed to be clean.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, April 14, 2008 5:00 PM To: freeswitch-users@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, April 14, 2008 4:33 PM To: freeswitch-users@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@mydomain.com <mailto:ext%2B1234@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@mydomain.com <mailto:ext%2B1234@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@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: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@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> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean. _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.
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@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
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, April 14, 2008 5:00 PM To: freeswitch-users@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, April 14, 2008 4:33 PM To: freeswitch-users@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@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@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@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@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>
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@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.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@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. _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
-
Ali Jawad
-
Anthony Minessale
-
jabber client
-
Michael Jerris