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FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue...
by Moiz Chinoy 19 Feb '09

19 Feb '09

Hi, I have downloaded and build the Freeswitch from http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. Everything built successfully. Then I configured PocketSphinx as described at http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. The problem is that prompts (wave files) are not being played properly while testing the Pizza demo i.e it plays, stops then start playing again... I am using heaset for testing. Moreover, the recognition seems to be very poor. Any clue what might be the issue? Moiz Chinoy. Following is snippet from log: 2009-02-18 16:59:59 [DEBUG] switch_core_session.c:513 switch_core_session_perform_receive_message() Send signal sofia/internal/1000(a)192.168.16.63 [BREAK] 2009-02-18 16:59:59 [NOTICE] mod_spidermonkey.c:2041 session_answer() Channel [sofia/internal/1000(a)192.168.16.63] has been answered 2009-02-18 16:59:59 [DEBUG] switch_channel.c:179 switch_channel_audio_sync() sofia/internal/1000(a)192.168.16.63 receive message [AUDIO_SYNC] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)192.168.16.63 entering state [completed] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)192.168.16.63 entering state [ready] 2009-02-18 16:59:59 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-02-18 16:59:59 [DEBUG] switch_core_io.c:654 switch_core_session_write_frame() sofia/internal/1000(a)192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:00 [INFO] switch_rtp.c:1422 rtp_common_read() Auto Changing port from 127.0.0.1:49194 to 192.168.16.63:49194 2009-02-18 17:00:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:11 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 2 seconds 1 heartbeat (core) 2009-02-18 17:00:18 [DEBUG] switch_core_media_bug.c:284 switch_core_media_bug_add() Attaching BUG to sofia/internal/1000(a)192.168.16.63 2009-02-18 17:00:18 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000(a)192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:51 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 20 seconds 1 heartbeat (core) 2009-02-18 17:00:51 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-02-18 17:00:58 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:58 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000(a)192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:00 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: ‎‎‎‎««««««««î‏î‏, Score: 100 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="100"> <result name="match">‎‎‎‎««««««««î‏î‏</result> <input>‎‎‎‎««««««««î‏î‏</input> </interpretation> 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [‎‎‎‎««««««««î‏î‏] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/70 2009-02-18 17:01:00 [INFO] js_modules/SpeechTools.jm:150 console_log() ----‎‎‎‎««««««««î‏î‏ 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ‎‎‎‎««««««««î‏î‏ =~ [Delivery:::Delivery] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ‎‎‎‎««««««««î‏î‏ =~ [Takeout:::Pickup] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ‎‎‎‎««««««««î‏î‏ =~ [Pickup:::Pickup] 2009-02-18 17:01:02 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:01:02 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000(a)192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:03 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2009-02-18 17:01:06 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recogniti… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Default IVR action
by Nandy Dagondon 19 Feb '09

19 Feb '09

hi everybody, i'm looking for a default action in an IVR if the caller doesn't press any key. for example, the caller will be transferred to the operator (or fifo) if no key is received after, let's say 5 seconds. is this available in the IVR? pls show a sample. tks, -nandy

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Re: [Freeswitch-users] Missing file for 1.0.3
by Michael S Collins 19 Feb '09

19 Feb '09

Sent from my iPhone On Feb 18, 2009, at 6:00 PM, Philip Patterson <philip.patterson(a)gmail.com > wrote: > Hi All. > > Have a fresh server and going to install FS on it. Went to the > download page (http://wiki.freeswitch.org/wiki/Installation_Guide) > and tried to download the "Phoenix" build, which is supposed to be > found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but > that file is nowhere to be found. Did the Wiki get updated before > the file was uploaded, or is there something else going on? Oops, my bad. That's exactly what happened. The file is actually 1.0.3RC1.tar.gz, although 1.0.3 should hit the server in the next day or so. Stay tuned! -MC > > > Philip > _______________________________________________ > 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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Missing file for 1.0.3
by Philip Patterson 19 Feb '09

19 Feb '09

Hi All. Have a fresh server and going to install FS on it. Went to the download page (http://wiki.freeswitch.org/wiki/Installation_Guide) and tried to download the "Phoenix" build, which is supposed to be found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but that file is nowhere to be found. Did the Wiki get updated before the file was uploaded, or is there something else going on? Philip

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Originate and bridge with lua
by Nik Middleton 18 Feb '09

18 Feb '09

Hi Guys, It's not clear from the docs how I can originate a call from within an lua script This what works in js, Question. How do I instantiate a new session, do I use the execute to dial, and same for bridge? Regards, if (!first_session.ready()) var new_session = new Session(tdial-string); if (!first_session.ready()) { disp_call(DROP) exit(); new_session.answer(); if (new_session.ready()) { bridge(first_session, new_session); }

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Skypiax on OS X
by Ivan C Myrvold 18 Feb '09

18 Feb '09

s it possible to run Skypiax on OS X? The wiki says Linux and Windows, but says nothing about OS X. I have been running FreeSWITCH on OS X for a couple of years now, and love it. Adding Skype gateway would be really sweet. Are there any plans for adding Skypiax to trunk, or do we have to build it only from svn branch? Ivan

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Re: [Freeswitch-users] Sending media streams to a media gateway
by Justin Miller 18 Feb '09

18 Feb '09

Anyone else have any ideas on this? -----Original Message----- From: fax(a)virgintechnologies.com [mailto:fax@virgintechnologies.com] Sent: Tuesday, February 17, 2009 05:45 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge <action application="set" data="bypass_media=true/> <action application="bridge" data="sofia/internal/someuser@somehost"/> that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, <fax(a)virgintechnologies.com> wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ 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 mailto:MSN%3Aanthony_minessale@hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888@conference.freeswitch.org http://iax:guest@conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888@conference.freeswitch.org pstn:213-799-1400

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Anyone running FS from a Thumb Flash USB?
by EdPimentl 18 Feb '09

18 Feb '09

Hello FS Members, Are there any example of FS running on a Thumb Flash USB? Thanks in advance, -E

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call goes to wrong context
by kokoska rokoska 18 Feb '09

18 Feb '09

Hi all, I have just "upgraded" to current trunk (before an hour or so), configuration remain the same (served through mod_xml_curl), but something has changed and I don'nt know "where", "what" and "why" :-) What's going on: I have few sofia profiles and each of them has its own context. When call arrives, http POST is made by FreeSWITCH to get the dialplan, but Caller-Context and Hunt-Context variables are always set to "default" regardless what contex I set in sofia profile the call comes in through. May be I miss something, but really have no idea where :-) Any hint is very appreciated. Best regards, kokoska.rokoska

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ipauth - directory
by Apostolos Pantsiopoulos 18 Feb '09

18 Feb '09

I noticed an "ip=" setting in the brian.xml sample file. The comments state that this is used for ipauth (IP based authentication?) What exactly is this setting. I cannot find anything in the wiki about it. Does it replace the use of the <param name="auth-calls" value="false"/> + ACL mechanism for IP authentication? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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