April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hangup and sound question
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I two a few questions about hangup and sounds: 1. In my dialplan, I have <extension name="catchall"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="default_language=zh"/> <action application="phrase" data="call-back-later"/> <action application="hangup" data="SUCCESS"/> </condition> </extension> After it play the call-back-later marco, it then goes on to play the "The extension you are dialing is unavailable..." even I have specified the HANGUP application. How come? How to remove that greeting? Also, when can i change the "The extension you are dialing is unavailable..." greeting to a localized version? 2. I am writing my application in javascript. Is there anyway to play sounds that are loaded from DB? Two ways I can think of are: 1. Use session.streamfile: but I need to extract the wav file from DB file and then stored it in the filesystem. Is it possible to do that in Javascript or I need to use something like PHP? 2. Sound marco: in the sound marco, is there a way to specify the file to be extracted from DB? 3. Let's say i have an external TTS engine. How do I make a call to it in javascript? The TTS engine can take text-message as input and generate wav content as STDOUT. What would be the best way to incorporate it into my application? 4. Is there anyway to display the post request within the console whenever the post is done?I tried curl_xml debug_on but it only gives me the received xml.

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freeswitch does not listen on the SIP port?
by Krzysiek 20 Apr '08

20 Apr '08

Hi there I've installed trixswitch lately and i have a problem. I haven't changed anything in the configuration files in the conf directory yet. I see that freeswitch daemon is running. But I see that it isn't listening on the SIP Port 5060? Do I have to do something special to force the Freeswitch to listen on that port? TIA for any help Chris

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voicemail_mod and spidermokey_odbc users
by Pete Kay 19 Apr '08

19 Apr '08

Hi, There are a few problems I found with voicemail and I am not sure if I did something wrong: Problem 1: With is my dialplan: <action application="set" data="default_language=zh"/> <action application="voicemail" data="default $${domain} ${dialed_ext}"/> This is the result: 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:115 switch_ivr_phrase_macro() No language specified - Using [zh] 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:256 switch_ivr_phrase_macro() Handle say:[] (zh:zh) 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:256 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-not_avaliable.wav] (zh:zh) 2008-04-19 22:12:35 [ERR] mod_sndfile.c:164 sndfile_file_open() Error Opening File[/usr/local/freeswitch/sounds/zh/cn/callie/voicemail/vm-not_avaliable.wav] [System error : No such file or directory.] 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:115 switch_ivr_phrase_macro() No language specified - Using [zh] 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:256 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (zh:zh) 2008-04-19 22:12:35 [DEBUG] mod_sndfile.c:170 sndfile_file_open() Opening File[/usr/local/freeswitch/sounds/zh/cn/callie/voicemail/8000/vm-goodbye.wav]rate 22050hz 2008-04-19 22:12:35 [WARNING] switch_core_file.c:114 switch_core_perform_file_open() Sample rate doesn't match 2008-04-19 22:12:35 [DEBUG] switch_ivr_play_say.c:860 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-04-19 22:12:35 [NOTICE] switch_resample.c:77 switch_resample_create() Activate Resampler 22050->8000 0.362812 2008-04-19 22:12:36 [DEBUG] switch_ivr_play_say.c:1121 switch_ivr_play_file() done playing file 2008-04-19 22:12:36 [NOTICE] switch_core_state_machine.c:198 switch_core_standard_on_execute() Hangup sofia/default/ 1004@192.168.1.104:5060 [CS_EXECUTE] [NORMAL_CLEARING] For voicemail/vm-not_avaliable.wav, it goes to /usr/local/freeswitch/sounds/zh/cn/callie/voicemail/ dir to fetch the file but for voicemail/vm-goodbye.wav, it goes to /usr/local/freeswitch/sounds/zh/cn/callie/voicemail/8000/ dir. Following the en sample, I put my wav files under the /8000/ only. With language=en, it always go to the <...>/8000/ dir, but with language=zh, it is not. How come? Also, where is the /8000/ defined? In my zh.xml: <include> <language name="zh" sound-path="$${base_dir}/sounds/zh/cn/callie" tts-engine="cepstral" tts-voice="callie"> <X-PRE-PROCESS cmd="include" data="demo/demo.xml"/>  <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/> </language> </include> In my zh.xml, the sound-path is /sounds/zh/cn/callie/, so I am not sure where the 8000 comes from? Problem 2: This time I change default_language to en: <action application="set" data="default_language=en"/> <action application="voicemail" data="default $${domain} ${dialed_ext}"/> when dialing in, the voice says something like " please record your message .... press any key to end the recording. I found that it will actually wait for a while before accepting my dtmf ( the "pressing any key to end recording"). I think the wait time is too long, cuz some people may just want to press the # key to hangup. With the current voicemail, it will repeat the "please record your message...." again and again. Is there a config for turning off the this waiting period? In the wiki ( http://wiki.freeswitch.org/wiki/Mod_voicemail) , there are a number of configuration parameter "keys", where can I find out more on how to use those keys and the available values that can be used? Problem 3: If I want to add something like " The person at extension XXXX " before the vm-not_available.wav, how to do it? Problem 4: When I tried to load the spidermonkey mod, I am getting the following error at startup: 2008-04-19 23:52:30 [ERR] mod_spidermonkey.c:930 sm_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: SQLFetch** When making DB call in javascript, I get the following error: 2008-04-20 03:15:09 [DEBUG] switch_core_state_machine.c:142 switch_core_standard_on_execute() sofia/default/1004@192.168.1.104:5060Execute javascript(switch.js) 2008-04-20 03:15:09 [ERR] mod_spidermonkey.c:3135 js_api_use() Error loading ODBC 2008-04-20 03:15:09 [ERR] switch.js:7 mod_spidermonkey() ReferenceError: ODBC is not defined Thank you very much for your kind help and input. Regards, Pete

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few problems when running the sample voicemail.js
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I am encountering a number of problems when trying to run the voicemail.js provided as one of the samples: 1. the etpan module is not available and I can't find the etpan module in the module.conf, so I have to comment out the below line. use("etpan"); So, what is the usage of the etpan? Will the voicemail.js work without it? 2. I don't know why the mkdir command below fails. I am already running fs as root. Any idea why? fsLogger("account dir = " + vmDir + "/" + vmAccount+"\n"); if(!dir.isDirectory){ vmDir.mkdir(vmDir + "/" + vmAccount); } 2008-04-19 03:26:28 [DEBUG] mod_spidermonkey.c:3132 js_api_use() Loading TeleTone 2008-04-19 03:26:28 [DEBUG] inc_logger.js:6 console_log() >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> account dir = /var/spool/freeswitch/10000 >>>>>>>>> undefined 2008-04-19 03:26:28 [ERR] answermachine.js:52 mod_spidermonkey() Error: File operation mkdir failed Thank you very much for your inputs. Regards, Pete

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Question with curl_xml
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I place my xml response generator directory.php under /var/www/fs/directory.php In my xml_curl_conf.xml, I have <binding name="directory fetcher"> <param name="gateway-url" value="http://localhost/fs/" bindings="directory.php"/> </binding> 2008-04-18 22:23:17 [ERR] mod_xml_curl.c:251 do_config() Binding has no url! 2008-04-18 22:23:17 [NOTICE] mod_xml_curl.c:273 do_config() Binding [directory fetcher] XML Fetch Function [http://localhost/fs/] [directory.php] 2008-04-18 22:23:17 [CONSOLE] switch_loadable_module.c:749 switch_loadable_module_load_file() Successfully Loaded [mod_xml_curl] 2008-04-18 22:23:17 [NOTICE] switch_loadable_module.c:245 switch_loadable_module_process() Adding API Function 'xml_curl' 2008-04-18 22:23:17 [CONSOLE] switch_loadable_module.c:749 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2008-04-18 22:23:17 [CONSOLE] switch_loadable_module.c:749 switch_loadable_module_load_file() Successfully Loaded [mod_event_socket] 2008-04-18 22:23:17 [NOTICE] switch_loadable_module.c:225 switch_loadable_module_process() Adding Application 'socket' 2008-04-18 22:23:17 [CONSOLE] switch_loadable_module.c:749 switch_loadable_module_load_file() Successfully Loaded [mod_iax] 2008-04-18 22:23:17 [NOTICE] switch_loadable_module.c:139 switch_loadable_module_process() Adding Endpoint 'iax' 2008-04-18 22:23:17 [NOTICE] sofia.c:1373 config_sofia() Adding Alias [ 192.168.1.104] for profile [default] 2008-04-18 22:23:17 [NOTICE] sofia.c:1383 config_sofia() Started Profile default [sofia_reg_default] 2008-04-18 22:23:17 [NOTICE] switch_scheduler.c:166 switch_scheduler_task_thread() Starting task thread 2008-04-18 22:23:17 [NOTICE] sofia.c:1383 config_sofia() Started Profile nat [sofia_reg_nat] 2008-04-18 22:23:18 [ERR] mod_xml_curl.c:190 xml_url_fetch() Received HTTP error 404 trying to fetch http://localhost/fs/ data: [hostname=ser&section=directory&tag_name=domain&key_name=name&key_value=192.168.1.104&domain= 192.168.1.104] <.....> freeswitch@ser> 2008-04-18 22:24:26 [ERR] mod_xml_curl.c:190 xml_url_fetch() Received HTTP error 404 trying to fetch http://localhost/fs/ data: [hostname=ser&section=directory&tag_name=domain&key_name=name&key_value=192.168.1.104&action=sip_auth&sip_profile=default&sip_user_agent=Linksys/SPA3102-3.3.6(GW)&sip_auth_username=1006&sip_auth_realm=192.168.1.104&sip_auth_nonce=25f4793c-0d53-11dd-bc54-07f6ea524331&sip_auth_uri=sip%3A192.168.1.104&sip_auth_qop=auth&sip_auth_cnonce=36fbccf&sip_auth_nc=00000001&sip_auth_response=e55890e6e780c28aa2ac5487c3e75905&sip_auth_method=REGISTER&key=id&user=1006&domain=192.168.1.104&ip= 192.168.1.103] 2008-04-18 22:24:26 [ERR] mod_xml_curl.c:190 xml_url_fetch() Received HTTP error 404 trying to fetch http://localhost/fs/ data: [hostname=ser&section=directory&tag_name=domain&key_name=name&key_value=192.168.1.104&action=sip_auth&sip_profile=default&sip_user_agent=Linksys/SPA3102-3.3.6(GW)&sip_auth_username=1007&sip_auth_realm=192.168.1.104&sip_auth_nonce=25fd93f0-0d53-11dd-bc54-07f6ea524331&sip_auth_uri=sip%3A192.168.1.104&sip_auth_qop=auth&sip_auth_cnonce=d14a908&sip_auth_nc=00000001&sip_auth_response=b1c78c65639c34ad74af62d8d5f9cd7e&sip_auth_method=REGISTER&key=id&user=1007&domain=192.168.1.104&ip= 192.168.1.103] I think this may just be a very minor error in my setup. Can someone please tell me what I did wrong? Thank you very much in advance for your help. Pete

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Question with marco speak
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I am having problem with the speak marco, can someone please help me? In the sample code: session.sayPhrase("speak", "Please leave a message after the beep", "en" ); Inside /usr/local/freeswitch/conf/lang/en/vm/sounds.xm, I added: <macro name="speak"> <input pattern="(.*)"> <match> <action function="play-file" data="voicemail/vm-hello.wav"/> </match> </input> </macro> I am getting can't find marco speak error: 2008-04-19 03:40:20 [ERR] switch_ivr_play_say.c:196 switch_ivr_phrase_macro( ) can't find macro speak. 2008-04-19 03:40:20 [WARNING] switch_ivr_play_say.c:330switch_ivr_phrase_macro () macro [speak] did not match any patterns 2008-04-19 03:40:20 [DEBUG] inc_logger.js:6 console_log() >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> session.streamFile rtn=[undefined] >>>>>>>>> undefined What did I do wrong? Thanks in advance for all your help. Regards, Pete

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newbie dialplan question
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I am working on some test on seeing how I can port my exist Asterisk stuff to Freeswitch. I am just getting started and I am hoping someone can give me some help to get started. I installed with all the default config and xml setting. Then, I bring up two SiP clients - one in the same machine as freeswitch (192.168.1.104) and the other one on another machine ( 192.168.1.102). When I dial an extension ( 1001, or 1002... etc ) from my SIP client on 192.168.1.102, I can make the call to the other client no problem. 2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1002@192.168.1.104:5060[9e6d6146-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:09:12 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->81001@default 2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@192.168.1.104:5061[9e93e564-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:09:12 [NOTICE] sofia.c:1603 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@192.168.1.104:5061! However, when I dial extension from the other SIP client, the one on 192.168.1.104, the call can't be routed. 2008-04-18 00:10:34 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@192.168.1.104:5061[cf710b58-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:10:34 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1001->1002@public 2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198 switch_core_standard_on_execute() Hangup sofia/default/ 1001@192.168.1.104:5061 [CS_EXECUTE] [NORMAL_CLEARING] 2008-04-18 00:10:34 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 20 (sofia/default/ 1001@192.168.1.104:5061) Ended It seems like the the call is being routed to the wrong context. How come this happens? I am using the standard default config xml files. Can anyone please help me? With freeswitch, is there anyway to debug/trace the processing of the call so I can see which condition it is in, and where it is routed to? That way, I can debug the config easier? Thanks alot for all your inputs and help. Regards, Pete

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multi-language voice
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I have a questions about multi-language voice support in FS. How to I add a new module such as mod_fr.so? I can make the necessary wav files, but how to create the modules? Thank you very much for your inputs. Regards Pete

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Build problem on Debian slice
by Anthony Knight 18 Apr '08

18 Apr '08

I'm having a problem building freeswitch on a xen debian etch slice (Details below). Everything from the standard modules.conf builds except for mod_spidermonkey. Here are the details of the build error. making all mod_spidermonkey cd config; make -j1 export cd pr; make -j1 export cd include; make export cd md; make export cd private; make export cd obsolete; make export cd src; make export cd io; make export cd malloc; make export cd md; make export cd unix; make export os_Linux_x86_64.s: Assembler messages: os_Linux_x86_64.s:46: Error: bad register name `%rdi)' os_Linux_x86_64.s:61: Error: bad register name `%rdi)' os_Linux_x86_64.s:75: Error: bad register name `%rdi)' os_Linux_x86_64.s:89: Error: bad register name `%rdi)' make[10]: *** [os_Linux_x86_64.o] Error 1 make[9]: *** [export] Error 2 make[8]: *** [export] Error 2 make[7]: *** [export] Error 2 make[6]: *** [export] Error 2 make[5]: *** [/usr/src/freeswitch-1.0/libs/js/libjs.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_spidermonkey-all] Error 1 make[2]: *** [all-recursive] Error 1 Details of my environment: ~:cat /proc/version Linux version 2.6.18-xenU (root(a)host277.rimuhosting.com) (gcc version 4.1.1 20070105 (Red Hat 4.1.1-52)) #4 SMP Wed Oct 24 01:50:19 UTC 2007 ~:uname -a Linux xyz.xyz.com 2.6.18-xenU #4 SMP Wed Oct 24 01:50:19 UTC 2007 x86_64 GNU/Linux ~:uname -m x86_64 ~:gcc --version gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) from googling about I think this is the same problem documented here ( http://trac.callcc.org/ticket/407) where the environment is reporting ARCH=x86_64, but should really be ARCH=x86. Any suggestions on how to overcome this problem? Thanks Tony

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Segfault
by David Knell 17 Apr '08

17 Apr '08

Hi all - Just had an interesting FS segfault. switch_xml_free (called from mod_xml_cdr) recursed 3500+ times and, I guess, the thing ran out of stack space. Having a look at the core dump shows that it's freeing a valid XML CDR structure for a call which wasn't cleared for ages; FS sat in a loop playing a prompt. The net result would have been a CDR with a vast number of <application> lines, had the segfault not occurred. Fixes: (a) don't get in this situation in the first place; (b) rewrite switch_xml_free to eliminate recursion; (c) limit the number of children created in switch_ivr_generate_xml_cdr; - and probably some more. My favourite's (c) - any alternatives/objections? If not, I'll pop this with a patch in Jira. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

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