November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[freeswitch-users] answering machine detection
by jonathan augenstine 08 Nov '08

08 Nov '08

I see a description for answer confirmation via DTMF. But, is there any work in progress to implement an actual answering machine detection capability? Jonathan

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Freeswitch Build fails (SVN Revision: 10294)
by B Karthik 07 Nov '08

07 Nov '08

Hi, The build fails for the latest svn revision:10294. Error message is as follows - ./.libs/libfreeswitch.so: undefined reference to `switch_dso_open' ./.libs/libfreeswitch.so: undefined reference to `switch_dso_data_sym' ./.libs/libfreeswitch.so: undefined reference to `switch_dso_destroy' Operating System is Debian Etch. Thanks. B Karthik.

2 1

Recording through Default.xml not working
by Baskar 07 Nov '08

07 Nov '08

Hi, * Recoding is done through default.xml. For past 1 month recoding is working fine. Suddenly for past 3day recording is not working. I did not modify any thing in deafult .xml . * * My deafult.xml file* <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="Local_Extension"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> <action application="answer"/> <action application="info"/> <action application="export" data="Answer-State"/> <action application="log" data="INFO Answer-State [${Answer-State}]\n"/> <action application="record_session" data="$${base_dir}/recordings/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="set" data="answer-state=${Answer-State}"/> <action application="info"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=false"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@ $${domain}"/> <action application="info"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="effective_caller_id_number=$1"/> <action application="set" data="dialed_ext=$1"/> <action application="info"/> <action application="set" data="answer-state=${Answer-State}"/> <action application="export" data="effective_caller_id_number=true"/> <action application="export" data="answer-state"/> <action application="set" data="hangup_after_bridge=false"/>  <action application="record_session" data="$${base_dir}/recordings/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bridge" data="sofia/default/$1(a)172.20.191.228"/> </condition> </extension> <extension name="nb_conferences"> <condition field="destination_number" expression="^(30\d{2})$"> <action application="answer"/>  <action application="conference" data="$1@default"/> </condition> </extension> </context> </include> correct me were is wrong -- Warm Regards, N.Baskar

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Different problems with "pizza" demo on newer build.
by mszlazak＠aol.com 06 Nov '08

06 Nov '08

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:

3 6

RPID and PAI questions ...
by Shelby Ramsey 06 Nov '08

06 Nov '08

Hello, Couple of quick questions: -- How do I keep FS from sending RPID? And strip the header? And likewise from respecting it? -- How do I add a PAI header -- example here of what I thought would work -- <action application="set" data="sip_h_P-Asserted- Identity=<sip: +17133000522(a)4.71.122.214>" /> ... but it doesn't seem to work (proceeding action is to dial a resource outside of the box -- How do I strip PAI headers from incoming calls? Thanks for any help! Shelby

1 0

More codec playback issues
by Brian Wood 06 Nov '08

06 Nov '08

I'm doing some more testing with various codecs. I'm seeing some more weirdness with the selection of a codec by file extension. I'm testing this on FreeSWITCH trunk 10270. The following script works with the .wav and .gsm extension, but everything else seems to fail. Using lowercase names (such as .g722) fail when recording with a "[ERR] switch_core_file.c:66 switch_core_perform_file_open() Invalid file format [g722] for [/tmp/recording.g722]!". Which is fine, I can use uppercase, but then that plays back garbled audio. This script should beep, record until you stop speaking, beep again, pause, beep, playback, beep, hang up. var allDigits = ""; function on_dtmf(session, type, obj, arg) { try { if (type == "dtmf") { allDigits += obj.digit; console_log("info", "DTMF digit: " + session.name + " [" + obj.digit + "] len [" + obj.duration + "]\n\n"); } } catch (e) { console_log("err", e + "\n"); } return true; } // .gsm works, .wav works //filename = "/tmp/recording.gsm" //filename = "/tmp/recording.wav" // These all result in garbled playback //filename = "/tmp/recording.GSM" filename = "/tmp/recording.G722" /filename = "/tmp/recording.SPEEX" //filename = "/tmp/recording.LPC" //filename = "/tmp/recording.PCMU" //filename = "/tmp/recording.PCMA" //filename = "/tmp/recording.AMR" session.waitForAnswer(10000); session.execute("sleep", "2000"); console_log("info", "*****BEGIN*****"); session.execute("gentones", "%(500,0,800)"); rtn = session.recordFile(filename, on_dtmf, "", 240, 500, 3); session.execute("gentones", "%(500,0,800)"); console_log("info", "*****END*****"); session.execute("sleep", "2000"); session.execute("gentones", "%(500,0,800)"); session.streamFile(filename); session.execute("gentones", "%(500,0,800)"); session.execute("sleep", "1000"); session.hangup();

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Problem in Routing G729A Calls
by shehzad p 06 Nov '08

06 Nov '08

I have to route the inbound calls of G729A codec. Calls comes to my freeswitch with codec G729A (As "annexb=no" is set) But when i route calls to termination gateway, calls are dropped (because of "annexb=no " is not set) Why "annexb=no" is removed while i route the calls? How can I set "annexb=no'? (I am using javascript for routing the calls) Does following SDP variables can help me in solving above problem? How to use those variables? http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Warm thanks in advance... MSP -- View this message in context: http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.ht… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 5

Inband DTMF Problem
by Klaus Teller 06 Nov '08

06 Nov '08

Hi Folks, It's me again with a DTMF issue. Here is what's going on. I have a remote IVR and i'm writing some code to communicate with it, exercise it, and test it. The remote IVR is playing DTMF as inband. Right now, Freeswitch can detect all DTMF digits from the IVR. The interaction between my code and the IVR is essentially a combination of playing files and sending and reading DTMFs. The issue: whenever my code plays a file or send a DTMF digit, the following attempt to read DTMF fails. By failing i mean that there will be a spurious DTMF detected by Freeswitch that was not sent by the IVR. That is, if i have something like: sendDTMF("3"); getDigits("38373", "#"); // # is the termination character After executing the sendDTMF, getDigits fails because it immediately detects a pound. Yet this pound is not sent by the IVR. I can check this by having wireshark running both on the IVR's computer and on Freeswitch's computer. Any idea what may be causing this problem? Is there some sort of Echo going on or something like that? Thanks, Klaus. -- "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail

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avoiding moh for meta_app
by Woody Dickson 06 Nov '08

06 Nov '08

Hi, I tried to avoid Freeswitch from playing moh when the meta_app is executed by setting hold_music=silence: <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hold_music=silence" /> <action application="set" data="call_timeout=120" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="language=zh" /> <action application="set" data="ringback=$${us-ring}" /> <action application="bind_meta_app" data="1 a s execute_extension::record XML features"/> <action application="bind_meta_app" data="2 a a execute_extension::stoprecord XML features"/> <action application="bind_meta_app" data="3 a s execute_extension::att_xfer XML features"/> <action application="bridge" data="sofia/gateway/openser/1000" /> <action application="phrase" data="user_not_reg" /> However, when #1 is pressed, Freeswitch still attempts to play moh: 92.168.1.101 Processing meta digit '1' [execute_extension::record XML features] 2008-11-07 05:18:30 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/internal/1001(a)192.168.1.101 [BREAK] 2008-11-07 05:18:30 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/1001(a)192.168.1.101[BREAK] 2008-11-07 05:18:30 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/external/1000 [BREAK] 2008-11-07 05:18:30 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/1000 [BREAK] 2008-11-07 05:18:30 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/external/1000 Command Execute playback(local_stream://moh) 2008-11-07 05:18:30 [DEBUG] mod_local_stream.c:320 local_stream_file_open() Opening Stream [moh/8000] 8000hz How can I fix this problem? Thanks alot in advance for your help. Woody

3 3

Anybody tried with Trunk between asterisk and freeswitch...?
by sambasivarao Vemula 06 Nov '08

06 Nov '08

Hi, Any body tried with Trunkig between freeswitch and asterisk . If any body tried and its working fine . Please share the details. Regards Samba DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

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