August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

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>

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Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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freeswitch as a session border controller
by Robert McNaught 30 Mar '09

30 Mar '09

Hi, I have been reading the recent news articles about freeswitch. I am particularly interested in using it to replace a session border controller we have (kagoor voiceflow 1000). Has anyone actually used this in production as a session border controller for NAT traversal? Thanks Robert McNaught

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Inbound calls
by whudson05 26 Jan '09

26 Jan '09

Hi, I'm experiencing a very specific problem to do with inbound calls through a gateway. When either end of the phone call hangs up then the other end will remain connected as if still in the conversation. If anyone has any idea why this would occur it would be greatly appreciated. Thanks -- View this message in context: http://www.nabble.com/Inbound-calls-tp18694165p18694165.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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lumenvox issue
by Frederick Jabre 01 Nov '08

01 Nov '08

Having trouble with Lumenvox. Could be a config error. I had this working with an older build. Mod_lumenvox is loaded successfully but the pizza demo keeps hanging up on me with this error: 2008-07-14 10:21:47 [NOTICE] switch_ivr_async.c:1773 switch_ivr_detect_speech() Hangup sofia/internal/fred-sip(a)200.56.122.8[CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-14 10:21:47 [NOTICE] switch_core_session.c:802 switch_core_session_thread() Session 1 (sofia/internal/fred-sip(a)200.56.122.8) Ended 2008-07-14 10:21:47 [NOTICE] switch_core_session.c:804 switch_core_session_thread() Close Channel sofia/internal/ fred-sip(a)200.56.122.8 [CS_HANGUP]

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Freeswitch and XML-RPC
by Klaus Teller 18 Oct '08

18 Oct '08

Hi, It seems i can send commands to Freeswitch using XML-RPC. The only document i could find on this is: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC Given that i'm using Java, i was wondering what else (beside originating a call) can I do via XML-RPC. How do i send for instance DTMF read requests, how do i send a request to stream a file? Any other document i could check to know what i can do with this? I would think that most of the functions that are available through Javascript are available via XML-RPC. But how do i send these commands? Thanks, Klaus. -- Psssst! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03

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Freeswitch billing
by Cliconnect 24 Sep '08

24 Sep '08

Hi, I am looking for a good and affordable (OS ?) billing software to integrate with FS. Can anybody provide any information ? thanks Duan

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dialpaln
by Cliconnect 22 Sep '08

22 Sep '08

Hi, I can call extensions like 1000 with <extension name="extensions"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/doublenat/$1%voipclic.com "/> </condition> </extension> and I can call PSTN with <extension name="PSTN"> <condition field="destination_number" expression="^(\d+)$"> <action application="bridge" data="sofia/gateway/inphonex/$0"/> </condition> </extension> Both config being in the same default.xml dialplan but I must use one or the other config. What I have to do in order to have them working at the same time. In other words the same user can call PSTN and the extension. thanks Duan

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Khushil Dep invites you to their network on Ecademy
by khushil.dep＠gmail.com 21 Sep '08

21 Sep '08

Connect with me on Ecademy Hi I've added you as one of my contacts on Ecademy where successful business people grow their business. Click here to connect with me: http://www.ecademy.com/account.php?op=signup&xref=310185 Thank you, Khushil Dep khushil.dep(a)gmail.com This email was sent to freeswitch-users(a)lists.freeswitch.org To avoid receiving these emails in the future send an email from that address to invite-unsubscribe(a)ecademy.com or go to http://www.ecademy.com/blockinvites.php?e=freeswitch-users@lists.freeswitch… Email:support@ecademy.com Company Registration:3651083 VAT:718 0377 36

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Command working on console but not using RPC
by Adeel Ansari 12 Sep '08

12 Sep '08

The sched_hangup command is working, like charm, on console. But when I issue the command using RPC I am receiving error. Although, originate command is working through RPC. Below is how I am doing it, in java. ===== client.execute("freeswitch.api", new Object[]{"sched_hangup","+6 "+uuid); ===== Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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