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by Adeel Ansari 20 Aug '08

20 Aug '08

Console Output: ========== UUID: d94510ba-6da6-11dd-bf51-11eda3fec726 ARGS: 2008-08-19 12:25:26 [ERR] mod_sofia.c:1946 sofia_outgoing_channel() Invalid Profile 2008-08-19 12:25:26 [NOTICE] mod_sofia.c:2065 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-08-19 12:25:26 [ERR] switch_ivr_originate.c:956 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] 2008-08-19 12:25:26 [WARNING] switch_cpp.cpp:889 originate() Error Creating Outgoing Channel! [sofia/sip/gizmo/01160176905074(a)proxy01.sipphone.com] 2008-08-19 12:25:26 [ERR] switch_cpp.cpp:974 waitForAnswer() session is not initalized 2008-08-19 12:25:26 [ERR] switch_cpp.cpp:835 ready() session is not initalized 2008-08-19 12:25:26 [NOTICE] switch_cpp.cpp:1050 console_log() ***********Babies! 2008-08-19 12:25:26 [ERR] switch_cpp.cpp:529 hangup() session is not initalized 2008-08-19 12:25:26 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/internal/1001(a)192.168.253.101[CS_EXECUTE] [NORMAL_CLEARING] ============= What I should look for, and where? Actually, I am using a SIP client to make a call to a default user 1001. And using dialplan config, I hook my Java program. In my Java program I am doing as below, ======== CoreSession session = null; CoreSession newSession1 = null; try { session = new JavaSession(uuid); newSession1 = new JavaSession(); session.originate(newSession1, "sofia/sip/gizmo/ 01160176905074(a)proxy01.sipphone.com"); session.waitForAnswer(newSession1); if (newSession1.ready() && session.ready()) { freeswitch.bridge(session, newSession1); } freeswitch.console_log("notice", "***********Babies!\n"); newSession1.hangup(); } finally { if (session != null) { session.delete(); } } ============= Furthermore, below is my gizmo.xml ========= <include> <user id="user-idxxx">  <gateways> <gateway name="gizmo"> <param name="username" value="user-idxxx"/> <param name="password" value="passwordxxx"/> <param name="from-domain" value="proxy01.sipphone.com"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="proxy01.sipphone.com"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry_seconds" value="3600"/> <param name="supress-cng" value="true"/> </gateway> </gateways> <variables> <variable name="effective_caller_id_name" value="M Hazriq"/> </variables> </user> </include> ========== Please steer me in the right direction. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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how to say numbers in java script
by vidhya sagar dixit 19 Aug '08

19 Aug '08

Hi , I am trying to play balance of an account. how to do that in java script without using tts. var x= 100; it should say 100 i have tried to use following example but didn't work form me. function sayphrase(phrase, args) { console_log("sayphrase: phrase=[" + phrase + "] args=[" + args + "]\n"); var rtn = session.execute("phrase", phrase + "," + args); return(rtn); } if (session.ready()) { session.answer(); session.execute("sleep","1000"); sayphrase("msgcount", "10"); session.hangup(); } can anyone tell me the correct syntax please. -- Thanks and Regards Vidhya Sagar Dixit

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playing sound file without answering the call
by vidhya sagar dixit 19 Aug '08

19 Aug '08

Hi All, Is it possible to play a sound file without answering the call in free switch ? -- Thanks and Regards Vidhya Sagar Dixit

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1529 at domain2 calling 9992 or 9992 at domain1
by Lee JJ 19 Aug '08

19 Aug '08

>>* # 1529 at domain2<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>calling 9992 at domain1 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> * >* >freeswitch at localhost<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> 2008-08-19 17:55:19 [WARNING] >>sofia_reg.c: *>* 1247 sofia_reg_parse_auth() can't find user *>* >[1529 at 210.243.126.69<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] *>* >You must define >>a domain called '210.243.126.69' in your directory *> * and add a user with the id="1529" attribute *>* >and you >>must configure your device to use the proper domain in it's *>* authentication credentials. **Need more >>explanations. * >This message can't be more clear. You do not have a domain called 210.243.126.69 with user 1529 defined in >your directory config. Because the 1529 user is at domain2/ or inter2 context not belong to domain1. >If you wish to set the context on a user then you'll need to use the user_context variable on the user in the >domain. <variable name="user_context" value="somecontext"/> Check out the conf/directory/default/1000.xml >user in the default config. I already did that. <variable name="accountcode" value="1529"/> <variable name="user_context" value="inter2"/> Brian , Sorry the things seem confused. Firstly , if I want to use sip request uri working like "sip:xxxx@yyyy" , what should I make it work in dialplan ? <action application="transfer" data="$1 DNS SRV inter2"/> Secondly , since the user context does not have relation with sip domain , what's the correct way ? ps: I feel FS does have some structure advantages , after using OpenSer , Asterisk for a while.

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Re: [Freeswitch-users] Problem using originate
by B Karthik 19 Aug '08

19 Aug '08

Can someone help me on this please.... Thanks. B Karthik ---------- Forwarded message ---------- > From: "B Karthik" <carthick84(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 18 Aug 2008 00:53:59 +0530 > Subject: [Freeswitch-users] Problem using originate > Hi, > > With bridge app, Freeswitch is able to place calls to multiple extensions > simultaneously. But with originate, is it possible? If Yes, can anyone give > me an example. > > The timeout option with originate also does not seem to work correctly. > > Timeout is not working (rings like a normal call and does not terminate > within the given timeout) if i do something like this - > newSession.originate(session, "user/1001(a)192.168.1.1",5); > > However it works if another extension is added with a comma - > newSession.originate(session, "user/1001@192.168.1.1,user/9999@192.168.1.1 > ",5); > > In the second ex, It does not call 9999 but the call disconnects exactly > after 5 seconds. > > I also tried the same with event_socket using api commands directly. The > results are same. > > OS: Windows XP SP2 > Compiler: Visual Studio 2008 > > Thanks. > B Karthik > >

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max_forwards
by Jonas Gauffin 19 Aug '08

19 Aug '08

Are max_forwards being used?

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1529@domain2 calling 9992 or 9992@domain1
by Lee JJ 19 Aug '08

19 Aug '08

Hello : user 1529 at 210.243.126.72 calling 9992 at 210.243.126.69 210.243.126.72 suppose a 2nd domain "inter2" 210.243.126.69 is default domain , and sip profile "internal" Why user 1529 have to be existed in "210.243.126.69" sip profile ? My client is X-Lite , and the callee info just type the text in callee no . # 1529@domain2 calling 9992@domain1 >freeswitch@localhost> 2008-08-19 17:55:19 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user >[1529(a)210.243.126.69] >You must define a domain called '210.243.126.69' in your directory and add a user with the id="1529" attribute >and you must configure your device to use the proper domain in it's authentication credentials. Need more explanations. I also tried other cases # 1529@domain2 calling 9992@domain2 >freeswitch@localhost> 2008-08-19 18:02:00 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel >sofia/inter2/1529(a)210.243.126.72<sofia/inter2/1529(a)210.243.126.72>[110509d1-7ad4-48ce-8095-a45300f7ed8c] >2008-08-19 18:02:00 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1529->9992@inter2 FS seems take callee's domain as searching profile . Like before test case. # 1529@domain2 calling 9992 >freeswitch@localhost> 2008-08-19 18:07:42 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel >sofia/inter2/1529(a)210.243.126.72<sofia/inter2/1529(a)210.243.126.72>[ce959295-e1e1-46c1-b7fd-381f93ee7644] >2008-08-19 18:07:42 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1529->9992@inter2 FS regards callee is the same domain with caller . Thanks !!

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playing a sound file without answering the call
by vidhya sagar dixit 19 Aug '08

19 Aug '08

Hi All, Is it possible to play a sound file without answering the call in freeswitch ? -- Thanks and Regards Vidhya Sagar Dixit

2 1

session.originate requires existing call leg?
by James Green 19 Aug '08

19 Aug '08

I am attempting to programatically call out from FreeSWITCH to my office phone as a test and simply play some music using Javascript. session.originate() looks closest but the first argument is an existing session (A-leg). But what if I'm creating the A-leg of the call? Do I just: session = new Session(); session.originate(session, my_destination); ? Or does 'session' exist even though there are no channels connected and if so can it act as the A-leg? Thanks, James

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ANNOUNCEMENT: Weekly FreeSWITCH documentation conference call, Wed Aug 20, 1PM Central
by Michael Collins 18 Aug '08

18 Aug '08

Here's the current link: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_08_20 Please add your questions/concerns/comments, etc. -MC

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