August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Voice not work in bgapi
by msp 20 Aug '08

20 Aug '08

Hi all, I need to make calls using bgapi command, When I make call using bgapi, ring comes on my both softphones. After Accepting the ring, call is bridged but there is no voice on both ends of phones. Another problem is with Playback, Instead of dialing exten I put playback. When I run bgapi command for playback, Ring comes on the softphone, But again after accepting, there is no voice/sound of playback. Any clue.., Please reply for more info if required. Thanks Shehzad.

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Did the call connect?
by James Green 20 Aug '08

20 Aug '08

Taking the following code: new_session.originate(session, dialout); new_session.waitForAnswer(20000); At this point either the call connected or we timed out. I expected to be able to use new_session.ready() to determine if the call successfully answered but this also returns true if the call timed out. The circumstances in which it will return false are not documented on the wiki. session.ready() is apparently not a function. So how does one check that the originate() ended up with a connected call? Thanks again, James

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Getting 401 Unauthorized session
by Adeel Ansari 20 Aug '08

20 Aug '08

Hi all, I am getting 401 while trying to make a call from a sotphone to a hardphone through voip. My username and password is both correct. Can you see any problem in the way I am doing it? ======== 2008-08-20 15:16:21 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/external/01160176900000 2008-08-20 15:16:21 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state() Channel sofia/external/01160176900000 entering state [calling] 2008-08-20 15:16:21 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_r_invite] status [401][Unauthorized] session: sofia/external/01160176900000 2008-08-20 15:16:21 [DEBUG] sofia_reg.c:1098 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"124.82.137.118":username:pwd'. 2008-08-20 15:16:21 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/external/01160176900000 2008-08-20 15:16:21 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state() Channel sofia/external/01160176900000 entering state [calling] 2008-08-20 15:16:22 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_r_invite] status [401][Unauthorized] session: sofia/external/01160176900000 2008-08-20 15:16:22 [DEBUG] sofia_reg.c:1098 sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 'Digest:"124.82.137.118":username:pwd'. 2008-08-20 15:16:22 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_r_invite] status [904][Operation has no matching challenge ] session: sofia/external/01160176900000 2008-08-20 15:16:22 [DEBUG] sofia.c:200 sofia_event_callback() event [nua_i_state] status [904][Operation has no matching challenge ] session: sofia/external/01160176900000 2008-08-20 15:16:22 [DEBUG] sofia.c:2145 sofia_handle_sip_i_state() Channel sofia/external/01160176900000 entering state [terminated] 2008-08-20 15:16:22 [NOTICE] sofia.c:2561 sofia_handle_sip_i_state() Hangup sofia/external/01160176900000 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2008-08-20 15:16:22 [DEBUG] switch_channel.c:1362 switch_channel_perform_hangup() Kill sofia/external/01160176900000 [KILL] 2008-08-20 15:16:22 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/01160176900000 [BREAK] 2008-08-20 15:16:22 [DEBUG] switch_ivr_originate.c:1438 switch_ivr_originate() Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] 2008-08-20 15:16:22 [INFO] mod_dptools.c:1794 audio_bridge_function() Originate Failed. Cause: NORMAL_UNSPECIFIED =========== Thanks, -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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(no subject)
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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