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storing voicemessage and greeting in DB
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I am wondering if storing and retrieving voicemaessage / greeting from DB makes sense especially for a load balancing environment. Is there anyway to do it in freeswitch? If not, is there any way of using some ftp-style storing and retrieving? How is it done typically to have voicemail working under a load balancing situation? Thanks in advance for your input. Thanks, Pete

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Please give suggestion on determining if a DTMF is valid
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I have a need to extra a list of valid DTMF input from DB ( as it may sometimes chang). The way I would do it is to extract the list of valid DTMF from DB before session.answer(). When the caller enters the DTMF when the menu is played, I need to search to see if the input is within the the resultset. There are few ways that I can think of: Option 1: Do a search query whenever the user enter the DTMF to check if it is valide. Option 2: Traverse the resultset (setting cursor to beginning of the resultset) to see if the entered DTMF exists or not. Option 3: Store the entire resultset into a hash table before playing the menu. >From the performance perspective, which one is better? Is there a better way to do it? The problem is that I am already experience some delay when doing the first DB call to get the entire list of valid DTMF from DB. I just think further delay is not good. Is there any good way to search for an element within a resultset? Thank you in advance for your input. Regards, Pete

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Hangup Cause: UNALLOCATED
by Alois Komenda 21 Apr '08

21 Apr '08

Hello, I'm trying to place a call between two SIPp instances over FreeSWITCH. Before the connection is set up completeley FS sends a BYE to the caller. In the logs is see the hangup cause is "UNALLOCATED". That happens only if there is a pause after sending the ACK. Hangup sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [CS_EXECUTE] [UNALLOCATED] 2008-04-17 14:53:46 [DEBUG] switch_channel.c:1293 switch_channel_perform_hangup() Kill sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [KILL] 2008-04-17 14:53:46 [DEBUG] switch_core_session.c:619 switch_core_session_signal_state_change() Kill sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [BREAK] 2008-04-17 14:53:46 [DEBUG] switch_core_state_machine.c:405 switch_core_session_run() sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 Running State Change CS_HANGUP 2008-04-17 14:53:46 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofe r.de:5060) Calling Handler for State HANGUP 2008-04-17 14:53:46 [DEBUG] mod_sofia.c:258 sofia_on_hangup() Channel sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 hanging up, cause: UNALLOCATED If I the call is placed by a softphone the same behavoir occurs. What does that mean? Is it a bug in FS? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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seek suggestion on TTS option
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I have a question about TTS efficiency. I have some text that I need to read out when someone calls in and these texts do change. Here are two options I am thinking about: In my configuration, I would have DB and FS on different servers. DB server is SCSI. Option 1:Store the text in DB Within JS: 1.Get the text from DB 2. use system (command) to create a wav file on locate file system 3. Use playback to play the wav file Option 2: Pre-created the file using NFS ( at the same machine as DB) Within JS: 1. Get the filename from DB 2. Get the wav file from NFS 3. Use playback to play the wav file In a highly-loaded environment, which option would be preferable considering speed and efficiency? Is there other better option? Any input will be greatly appreciated. Thanks, Pete

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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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