November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_xml_curl and basic authentication
by MEHDi CHAABOUNi 10 Nov '08

10 Nov '08

Hi, I'm trying to fetch a dialplan using mod_xml_curl. The problem is that I can't get it to work when basic authentication is enabled in my webapp. I think that FS is not sending the Authentication in the header when making requests. Am I missing something? Is there something else that I should set beside adding the following line to the file "xml_curl.conf.xml" <param name="gateway-credentials" value="username:password"/> here's my "xml_curl.conf.xml": <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="dialplan fetcher"> <param name="gateway-url" value="http://localhost:8080/dialplan" bindings="dialplan"/> <param name="gateway-credentials" value="root:root"/> </binding> </bindings> </configuration> Thanks

2 3

How to find an extension in public.xml ?
by henkoegema 10 Nov '08

10 Nov '08

I have an offsite phone in ( as described in http://wiki.freeswitch.org/wiki/Example_Offsite_phones) //freeswitch/conf/dialplan/public.xml: ============================= <extension name="remote_extension_2014"> <condition field="destination_number" expression="^2014$"> <action application="bridge" data="sofia/doublenat/$1%$${domain}"/> </condition> </extension> When I dial 2014, FS looks through the whole file in ../dialplan/default.xml, but can't find it there (because it's not definded in default.xml) (http://pastebin.freeswitch.org/6062) Q: How can I force FS to look in ../dialplan/public.xml if it can't find an entry in default.xml ? Henk -- View this message in context: http://www.nabble.com/How-to-find-an-extension-in-public.xml---tp20418918p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

2 6

Where to contribute to Wiki for newbie example
by Gavin Henry 10 Nov '08

10 Nov '08

Hi All, With the help of bkw_ today in #freeswitch, I got a working conference setup for an inbound SIP call from our ITSP. Basically handling a call like sip:01224279494@our_fs_ip Steps are: 1. vi conf/autoload_configs/acl.conf.xml to create your new acls (called conferences) 2. vi conf/sip_profiles/internal.xml to add/change you acl name ( <param name="apply-inbound-acl" value="conferences"/> ) 3. vi conf/dialplan/public.xml to create your DID and where to send it (to your not yet created conference context) 4. cp conf/dialplan/default.xml conf/dialplan/conferences.xml; vi conf/dialplan/conferences.xml to define your new conference context and what conference to send the call to 5. vi conf/autoload_configs/conference.conf.xml (PIN and conf numbers) to define yoru actual conference There will be expanded with screens for wiki. Which page should this go on, as it can be expanded for anything a newbie wants to do to get started. Thanks, Gavin. -- http://www.suretecsystems.com/services/openldap/

1 0

Hangup problem/SIP BYE lacking authentication
by Wellie Chao 10 Nov '08

10 Nov '08

Hangups do not work for me under certain circumstances. Here is the background information: * Our carrier uses a Metaswitch server with Acme Packet in front as a proxy/SBC. Only the Acme Packet machine is publicly visible (64.115.128.6). * Our Freeswitch server is at 216.57.23.143. * For calls originating from Freeswitch and terminating on Metaswitch (216.57.23.143 -> 64.115.128.6), Freeswitch authenticates with Metaswitch and everything works hunky-dorey. Either side can hang up and the other side will automatically hang up without requiring a manual hang-up. Now the problem: For calls originating from Metaswitch to Freeswitch (64.115.128.6 -> 216.57.23.143), Metaswitch does not authenticate with Freeswitch. Metaswitch also does not use the existing authenticated registration that our Freeswitch server initiates with Metaswitch upon startup of Freeswitch. Metaswitch just begins a new (unauthenticated) session and we have configured Freeswitch to allow any inbound calls from 64.115.128.6 without requiring authentication. We receive inbound calls (Metaswitch to Freeswitch, 64.115.128.6 -> 216.57.23.143) just fine. The phone rings and we can have a normal conversation. If the caller (the endpoint attached to Metaswitch) hangs up, both sides hang up. If I hang up (remember, I'm at the endpoint attached to Freeswitch), the caller's line remains attached forever. I have recorded a packet trace. Look at freeswitch_2.cap in the ZIP file, and you want to graph the first call starting at 21.202 and ending 53.798. If you go to time 53.087, you can see that my Freeswitch server sends a BYE to Metaswitch. This is a result of me hanging up my phone. At time 53.089, you see Metaswitch responding with 401 Unauthorized. Later at time 53.777, you see a BYE from Metaswitch to Freeswitch, but you should ignore this because that was a result of the caller (the guy hooked up to Metaswitch) manually hanging up. If he had not hung up his phone, the BYE from Metaswitch to Freeswitch would not have been issued and his phone would just stay on the line forever. Also, when I hang up my phone, I see the following at the Freeswitch console: 2008-10-29 23:03:28 [ERR] sofia_reg.c:1089 sofia_reg_handle_sip_r_challenge() No Matching gateway found I presume that Freeswitch emits this error because it got the 401 Unauthorized from Metaswitch. I also asked our carrier for a packet trace of a successful hangup on the Aastra platform (the engineer at the carrier says it is an Asterisk derivative -- I'm not sure about that). Look at Aastra_authentication_test.cap in the ZIP file. Graph the first call starting at 43.633 and ending 93.156. If you go to 93.118, you'll see that the Aastra server sends a BYE. Just like our Freeswitch scenario, Metaswitch sends back a 401 Unauthorized, but in response to the 401 Unauthorized, Aastra then sends back another BYE with the difference that the second BYE is authenticated. Metaswitch gets the second BYE and responds with 200 OK. I am pretty sure that if Freeswitch were to send back a second BYE (but with authentication), it would work fine. Now my question is how can I do this? I am not sure if this divergence of behavior is caused by: (a) my own error in configuring Freeswitch, (b) Metaswitch lacking standard SIP support (maybe it's not supposed to send the 401 Unauthorized), or (c) Freeswitch lacking standard SIP support (maybe it's supposed to send back a second BYE with authentication automatically). I don't know the SIP standards (or Freeswitch) well enough to know whether this problem is caused by me or by a deficiency in one of the two products (Metaswitch or Freeswitch). Can you provide some pointers? The ZIP file with the packet traces can be downloaded here: http://216.57.23.143/hangup_problem.zip

4 17

Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.
by mszlazak＠aol.com 10 Nov '08

10 Nov '08

Hi Anthony and Brian. I should have done this yesterday but I'm learning debugging in this IDE environment as I go. Anyway, have you looked at the .lm grammar files in the "pizza demo"? For the grammar file download in the wiki, I finding 9 .lm files: 6859, pizza_arso, pizza_crust, pizza_order, pizza_size, pizza_specialty, pizza_toppings, pizza_type, pizza_yesno. Of these, 4 files are empty: pizza_crust, pizza_size, pizza_specialty, pizza_toppings. I commented out the related sections in ps_pizza.js and the "pizza demo" seemed to work in this shortened form. It took the type of order (nothing else), order confirmation, when to come pick-up and then hung-up. Could I get the rest of the .lm files? Updating the grammar files download for "pizza" would nice. Thanks. Mark. -----Original Message----- From: Anthony Minessale <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thu, 6 Nov 2008 6:22 am Subject: Re: [Freeswitch-users] Different problems with "pizza" demo on newer build. can you please download and build the development snapshot instead of the 1.0.1 since we are almost to 1.0.2 now. http://files.freeswitch.org/freeswitch-snapshot.tar.gz On Wed, Nov 5, 2008 at 10:05 PM, <mszlazak(a)aol.com> wrote: Yup, I'm using ps_pizza.js. Mark -----Original Message----- From: Brian West <brian(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Wed, 5 Nov 2008 4:50 pm Subject: Re: [Freeswitch-users] Different problems with "pizza" demo on newer build. Looks like the score is messed up. Are you using ps_pizza.js? /b On Nov 5, 2008, at 6:44 PM, mszlazak(a)aol.com wrote: OK, I did my first FS "build" today from download "freeswitch-1.0.1.tar.gz" on Windows XP Pro with Visual C++ 2008 Express instead of using my previous MSI installed build which apparently uses the older FS version 1.0.9570. I'm not getting a crash as I did before with the MSI build on the "pizza" demo. Instead, in the newer build, the demo gets stuck in a loop and can't break out no matter what I say. See below: Any advice? Thank you in advance. Mark. </interpretation> 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [²²²²½½½½½½½½ε■ε■] 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:21 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:21 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:25 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:25 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:26 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:26 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:27 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:27 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:33 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:33 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:34 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:35 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:37 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:37 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:42 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:42 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:43 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: DELIVERY, Score: 0 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">DELIVERY</result> <input>DELIVERY</input> </interpretation> 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [DELIVERY] 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:44 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:44 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:50 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:50 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:50 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:51 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:51 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:51 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:53 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:53 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:58 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:58 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:59 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:23:00 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:23:00 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:23:01 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: PICKUP, Score: 0 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">PICKUP</result> <input>PICKUP</input> </interpretation> 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [PICKUP] 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:23:03 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:23:03 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:23:04 [DEBUG] sofia.c:197 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/internal/1000(a)10.0.0.2 2008-11-05 16:23:04 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [calling] 2008-11-05 16:23:04 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [ready] 2008-11-05 16:23:04 [DEBUG] sofia.c:2133 sofia_handle_sip_i_state() Remote SDP: Instant access to the latest & most popular FREE games while you browse with the Games Toolbar - Download Now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org = _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Instant access to the latest & most popular FREE games while you browse with the Games Toolbar - Download Now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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SIP-Request firstline
by Helmut Kuper 10 Nov '08

10 Nov '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, is there a way in dialplan or in application api to set the host part only the firstline in SIP-INVITE-Requests without setting the hostpart in To header field? If not, is there a way to simply forward the first INVITE to a sip-proxy, where dirstline's host part and to header host part are different? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkkYDgkACgkQ4tZeNddg3dw0QgCgw4ayQR8lnE8i9eUKID92Mc8e /AUAn2Y82M6eO34NZc6W1q0HwUsQ4fSu =5Nop -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] Question on using java_mod
by I Ki 10 Nov '08

10 Nov '08

Ah, so I should not compile the PhoneTest.java to PhoneTest.class only but jar-file instead? This is what I did: - verified my classpath in the java.conf.xml: <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/> - my PhoneTest.class is located in /usr/local/freeswitch/script - directory, same as where freeswitch.jar is located - I created PhoneTest.jar file from PhoneTest.class - changed my dialplan to: <action application="java" data="/usr/local/freeswitch/scripts/PhoneTest.jar PhoneTest arg1 arg2 arg3"/> - and seems to be working!!! at least it was called this time and it initiated. Couple of small errors but I'm sure it's minor, will have a look at those later. Thanks a million, would have never figured it out by myself. Is there any technical reason why class-file cannot be called as an binary directly? Nic ------------------ No, not the path to *freeswitch.jar*, use the path to *your* JAR containing the class. Bye Damjan > > Thanks Klaus, > > Tried that as well, if I use a full path for PhoneTest then I get same > error as earlier. > > If I call only PhoneTest without any path > (/usr/local/freeswitch/scripts/freeswitch.jar PhoneTest arg1 arg2..) > then I get: > > Exception in thread "Thread-2" java.lang.ClassNotFoundException: PhoneTest > at java.net.URLClassLoader$1.run(URLClassLoader.java:200) at > java.security.AccessController.doPrivileged(Native Method) at > java.net.URLClassLoader.findClass(URLClassLoader.java:188) at > java.lang.ClassLoader.loadClass(ClassLoader.java:306) at > java.lang.ClassLoader.loadClass(ClassLoader.java:251) at > java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319) at > java.lang.Class.forName0(Native Method) at > java.lang.Class.forName(Class.java:242) at > org.freeswitch.Launcher.launch(Launcher.java:65) > It's a same error message if I leave the full path off from > freeswitch.jar as well. There must be something what I have not > understood about the documentation here.. > > N > > > > > > You might want to specify not only the class name, but also its > package name as well as the jar file that contains it. <action application="java" > data="/path/to/file.jar fully.qualified.class.name arg1 arg2 arg3"/> > Klaus. > > From: i.mkmo(a)hotmail.comTo: freeswitch-users(a)lists.freeswitch.orgSubject: > Question on using java_modDate: Sun, 9 Nov 2008 21:28:23 +0800 > > Hey, I've been testing the usage of mod_java and reading Freeswitch > documentation but been having some problems which I think might even > be something very simple which I've missed. Hope some could give a > hint..mod_java installation has gone through successfully and > mod_java.so is compiled and installed. Configuration files > java.conf.xml and dialplan are configured as instructed in > http://wiki.freeswitch.org/wiki/JavaCompilation of the Java example in > the same page is successful (CLASSPATH included freeswtich.jar of course). > Problem comes when I try to initiate the script via dialplan - that is > I'm trying to make a call to an extension which then calls this > example script (dialplan says: <action application="java" > data="PhoneTest arg1 arg2 arg3"/>). Error message gives me: Exception in thread "Thread-1" > java.lang.ClassNotFoundException: /usr/local/freeswitch/scripts/PhoneTest > at java.lang.Class.forName0(Native Method) at > java.lang.Class.forName(Class.java:242) at > org.freeswitch.Launcher.launch(Launcher.java:65) I presume this means > that the script is called invalid manner? I've been trying to call > the PhoneTest with full path, locating it together with > freeswitch.jar, creating a directory PhoneTest just name > it.Documentation does not say if the Java binary needs to be in certain location or not - is there > amandatory location for it? What else am I doing wrong? My system is > running in CentOS 5, Java version JDK1.5.0_16 so the unstability described > in Wiki should not be a reason. Nic > > Enrich your blog with Windows Live Writer. Windows Live Writer > _________________________________________________________________ > NEW! Get Windows Live FREE. > http://www.get.live.com/wl/all________________________________________ > _______ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Easily edit your photos like a pro with Photo Gallery. http://get.live.com/photogallery/overview

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Re: [Freeswitch-users] Question on using java_mod
by I Ki 10 Nov '08

10 Nov '08

Thanks Klaus, Tried that as well, if I use a full path for PhoneTest then I get same error as earlier. If I call only PhoneTest without any path (/usr/local/freeswitch/scripts/freeswitch.jar PhoneTest arg1 arg2..) then I get: Exception in thread "Thread-2" java.lang.ClassNotFoundException: PhoneTest at java.net.URLClassLoader$1.run(URLClassLoader.java:200) at java.security.AccessController.doPrivileged(Native Method) at java.net.URLClassLoader.findClass(URLClassLoader.java:188) at java.lang.ClassLoader.loadClass(ClassLoader.java:306) at java.lang.ClassLoader.loadClass(ClassLoader.java:251) at java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319) at java.lang.Class.forName0(Native Method) at java.lang.Class.forName(Class.java:242) at org.freeswitch.Launcher.launch(Launcher.java:65) It's a same error message if I leave the full path off from freeswitch.jar as well. There must be something what I have not understood about the documentation here.. N You might want to specify not only the class name, but also its package name as well as the jar file that contains it. <action application="java" data="/path/to/file.jar fully.qualified.class.name arg1 arg2 arg3"/> Klaus. From: i.mkmo(a)hotmail.comTo: freeswitch-users(a)lists.freeswitch.orgSubject: Question on using java_modDate: Sun, 9 Nov 2008 21:28:23 +0800 Hey, I've been testing the usage of mod_java and reading Freeswitch documentation but been having some problems which I think might even be something very simple which I've missed. Hope some could give a hint..mod_java installation has gone through successfully and mod_java.so is compiled and installed. Configuration files java.conf.xml and dialplan are configured as instructed in http://wiki.freeswitch.org/wiki/JavaCompilation of the Java example in the same page is successful (CLASSPATH included freeswtich.jar of course). Problem comes when I try to initiate the script via dialplan - that is I'm trying to make a call to an extension which then calls this example script (dialplan says: <action application="java" data="PhoneTest arg1 arg2 arg3"/>). Error message gives me: Exception in thread "Thread-1" java.lang.ClassNotFoundException: /usr/local/freeswitch/scripts/PhoneTest at java.lang.Class.forName0(Native Method) at java.lang.Class.forName(Class.java:242) at org.freeswitch.Launcher.launch(Launcher.java:65) I presume this means that the script is called invalid manner? I've been trying to call the PhoneTest with full path, locating it together with freeswitch.jar, creating a directory PhoneTest just name it.Documentation does not say if the Java binary needs to be in certain location or not - is there amandatory location for it? What else am I doing wrong? My system is running in CentOS 5, Java version JDK1.5.0_16 so the unstability described in Wiki should not be a reason. Nic Enrich your blog with Windows Live Writer. Windows Live Writer _________________________________________________________________ NEW! Get Windows Live FREE. http://www.get.live.com/wl/all

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Conference (unable to unhold)
by Baskar 10 Nov '08

10 Nov '08

*Hi, api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228) ====>1001 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>**Both caller and extension will be transfer into conference room (1001 +9841799874 in conference room 3001 ** api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078(a)172.20.191.228) ====> 1001 call another caller 9962576232 api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>**Transfer only the caller 9841799874 to conference room 3001* * while calling the second no 9962576232 the first line get hold after the transfer of bleg they all come in to the conference room but the first caller is in hold we cant able to realse the call through api we have to manually go the soft phone and unhold the call . but we need the unhold should be done in api what is the solution. Can any one help where i am wrong. * -- Warm Regards, N.Baskar

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Improving FreeSwitch/PocketSphinx Recognition Accuracy With Signal Pre-Processing.
by mszlazak＠aol.com 10 Nov '08

10 Nov '08

I have recently read an article on improving recognition accuracy of any speech recognition system (http://www.google.com/patents?id=mZ2jAAAAEBAJ&dq=Pattern+recognition+accura…) where the voice input signal, x, has two modified copies made of itself before reaching the recognition engine. The 3 lists of possible utterances derived from each of these signals is looked at to determine the best choice for what the utterance could have been. This technique is claimed to reduce recognition errors up to 80% and obviously will slow down performance but is implemented in call centers that utilize Fluency Voice technology. The technique seems simple. Basically, one variant of the original voice signal, x, is "expanded" or amplified in a nonlinear way so that weaker signal component strengths are magnified more so than stronger ones. In the other variant, the signal x is expanded so that stronger signal component strengths are magnified to a greater extent than weaker ones. Also, there are gain factors applied to these new signals to account for overall signal strength changes. At least that's my take on the article. The formula used for signal expansion is just a simple power function of the form y = g*x^c, where y is the new signal, g is the gain factor and c is the the power. Expected values of c vary from 0.6 to 1.4 and g is around 20 for c=0.6 and g=0.1 for c=1.4 (See the cited article for more details). Now, this is just one example and there maybe better or other ways to do improve pocketsphinx (PS) accuracy for phone signals. So my interest is about ways one could do pre-processing of signals in FS before they "hit" pocketsphinx. I presume that the generated lists are an issues that PS folks would have to help me with. Any guidance on this and how to implement some form of signal pre-processing in FS would be much appreciated. Mark.

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