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Mod pocketsphinx in Free-switch not recognized vice
by Muhammad Abdullah 05 Dec '13

05 Dec '13

Hi Every one, Please help me where i am getting wrong in configuration , I have follow all steps that describe in http://wiki.freeswitch.org/wiki/Mod_pocketsphinx . Install Sound files and Grammar files also, Enable mod Pocketsphinx and java mode , I have not seen any error while loading Free-switch , but Pizza demo greeting listen when dialing a Demo number, but it can recognized my voice , But i have seen in log *detect_speech(pocketsphinx pizza_size undefined)* Here is my log .. 2013-12-04 14:07:14.223390 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->800800 in context default Dialplan: sofia/internal/1000@192.168.1.15:5060 parsing [default->pizza_demo] continue=false Dialplan: sofia/internal/1000@192.168.1.15:5060 Regex (PASS) [pizza_demo] destination_number(800800) =~ /^(800800)$/ break=on-false Dialplan: sofia/internal/1000@192.168.1.15:5060 Regex (PASS) [pizza_demo] ${module_exists(mod_spidermonkey)}(true) =~ /true/ break=on-false Dialplan: sofia/internal/1000@192.168.1.15:5060 Regex (PASS) [pizza_demo] ${module_exists(mod_pocketsphinx)}(true) =~ /true/ break=on-false Dialplan: sofia/internal/1000@192.168.1.15:5060 Action javascript(ps_pizza.js) 2013-12-04 14:07:14.223390 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1000@192.168.1.15:5060) State Change CS_ROUTING -> CS_EXECUTE 2013-12-04 14:07:14.223390 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.223390 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/1000@192.168.1.15:5060) State ROUTING going to sleep 2013-12-04 14:07:14.223390 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000@192.168.1.15:5060) Running State Change CS_EXECUTE 2013-12-04 14:07:14.223390 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/1000@192.168.1.15:5060) State EXECUTE 2013-12-04 14:07:14.223390 [DEBUG] mod_sofia.c:243 sofia/internal/ 1000@192.168.1.15:5060 SOFIA EXECUTE 2013-12-04 14:07:14.223390 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1000@192.168.1.15:5060 Standard EXECUTE EXECUTE sofia/internal/1000@192.168.1.15:5060 javascript(ps_pizza.js) 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5262 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3186 Set Codec sofia/internal/1000@192.168.1.15:5060 GSM/8000 20 ms 160 samples 13200 bits 2013-12-04 14:07:14.243351 [DEBUG] switch_core_codec.c:111 sofia/internal/ 1000@192.168.1.15:5060 Original read codec set to GSM:3 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:5430 Set 2833 dtmf send/recv payload to 101 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3445 AUDIO RTP [sofia/internal/1000@192.168.1.15:5060] 192.168.1.15 port 27068 -> 192.168.1.14 port 50264 codec: 3 ms: 2 0 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3445 AUDIO RTP [sofia/internal/1000@192.168.1.15:5060] 192.168.1.15 port 27068 -> 192.168.1.14 port 50264 codec: 3 ms: 2 0 2013-12-04 14:07:14.243351 [DEBUG] switch_rtp.c:2012 Starting timer [soft] 160 bytes per 20ms 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3712 Set 2833 dtmf send payload to 101 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3718 Set 2833 dtmf receive payload to 101 2013-12-04 14:07:14.243351 [DEBUG] sofia_glue.c:3745 sofia/internal/ 1000@192.168.1.15:5060 Set rtp dtmf delay to 40 2013-12-04 14:07:14.243351 [NOTICE] sofia_glue.c:4356 Pre-Answer sofia/internal/1000@192.168.1.15:5060! 2013-12-04 14:07:14.243351 [DEBUG] switch_channel.c:3364 (sofia/internal/ 1000@192.168.1.15:5060) Callstate Change RINGING -> EARLY 2013-12-04 14:07:14.243351 [DEBUG] mod_sofia.c:866 Local SDP sofia/internal/ 1000@192.168.1.15:5060: v=0 o=FreeSWITCH 1386167766 1386167767 IN IP4 192.168.1.15 s=FreeSWITCH c=IN IP4 192.168.1.15 t=0 0 m=audio 27068 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-12-04 14:07:14.243351 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.243351 [DEBUG] sofia.c:5752 Channel sofia/internal/ 1000@192.168.1.15:5060 entering state [completed][200] 2013-12-04 14:07:14.243351 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.243351 [NOTICE] mod_spidermonkey.c:2109 Channel [sofia/internal/1000@192.168.1.15:5060] has been answered 2013-12-04 14:07:14.283378 [DEBUG] switch_channel.c:3643 (sofia/internal/ 1000@192.168.1.15:5060) Callstate Change EARLY -> ACTIVE 2013-12-04 14:07:14.283378 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2013-12-04 14:07:14.345270 [DEBUG] switch_rtp.c:3727 Correct ip/port confirmed. 2013-12-04 14:07:14.443246 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.443246 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.443246 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1000@192.168.1.15:5060 [BREAK] 2013-12-04 14:07:14.463380 [DEBUG] sofia.c:5752 Channel sofia/internal/ 1000@192.168.1.15:5060 entering state [ready][200] 2013-12-04 14:07:16.563518 [DEBUG] switch_ivr_play_say.c:1722 done playing file /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-Greeting.wav *EXECUTE sofia/internal/1000@192.168.1.15:5060 <http://1000@192.168.1.15:5060> detect_speech(pocketsphinx pizza_size undefined)* 2013-12-04 14:07:16.563518 [DEBUG] switch_core_media_bug.c:532 Attaching BUG to sofia/internal/1000@192.168.1.15:5060 2013-12-04 14:07:17.723606 [DEBUG] switch_core_io.c:493 Setting BUG Codec GSM:3 EXECUTE sofia/internal/1000@192.168.1.15:5060 detect_speech(resume) 2013-12-04 14:07:18.003623 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2013-12-04 14:07:20.903377 [DEBUG] switch_ivr_play_say.c:1722 done playing file /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-Size.wav EXECUTE sofia/internal/1000@192.168.1.15:5060 detect_speech(resume) EXECUTE sofia/internal/1000@192.168.1.15:5060 detect_speech(resume) 2013-12-04 14:07:23.823564 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms Many Thanks Iffi

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Copy all sip headers between two legs
by Mickael Hubert 05 Dec '13

05 Dec '13

hi list, I want copy all sip headers between two legs, but I don't obtain a good result. example: P-Asserted-identity, P-Preferred-Identity, Diversion, etc ... aren't copied for leg B. I have found "sip_copy_custom_headers=true" but it's just for X-Header. Actually, I use this code below, but it is not easy :( *<condition field="${sip_P-Asserted-Identity}" expression="\+(\d+)" break="never"> <action application="log" data="LOG Modifying PAI sip:+33xxxxxxxxx@DOMAIN" /> <action application="export" data="sip_h_P-Asserted-Identity=<sip:+$1@3008LI.hexavoip.fr <1(a)3008LI.hexavoip.fr>>" />* Have you an idea for me ? Thanks in advance -- Cordialement HUBERT Mickaël Ingénieur VOIP - Hexanet --

1 1

Fax passthrough issues with FreeSWITCH
by John Platts 04 Dec '13

04 Dec '13

Some of our customers have reported problems sending and receiving faxes over VoIP through an analog telephone adapter with some models of fax machines. Some of our customers are using fax machines connected to the UC500 FXS port and some of our other customers are using SPA112 ATAs. For fax calls that use the G.711 codec, we need to be able to suppress the SG3 ANSam tone and force the fax machine to negotiate at a slower speed. We also need to be able to pass through the fax signal subsequent to negotiation while ensuring that the fax call does not get dropped. Is there a way to accomplish this goal in FreeSWITCH?

1 0

python in freeswitch
by Joli Martinez 04 Dec '13

04 Dec '13

Hello, I would like to be able to mute/unmute users in a conference using a python script. Not sure how to do that. Please help, Thanks

3 2

FreeSWITCH and Raspberry.
by Fernando Villarroel 04 Dec '13

04 Dec '13

Dear All. Someone tested FS and Raspberry as a SOHO PBX. I need to know their experiences, i want to run in Raspberry : USBfxo Sangoma. FreeSWITCH PostreSQL Nginx Django What do you think? I wait your feedback or advice. Enviado desde mi iPhone

7 14

Getting FreeSWITCH to use RTP port + 1 for RTCP
by James Mortensen 04 Dec '13

04 Dec '13

Hello, I've searched the list and documentation, and it's not yet clear if FreeSWITCH is able to send candidates in the same manner as we were doing with Asterisk 1.5 where it sends the RTCP to the RTP port +1. In other words, I'd see candidates that looked like this: a=candidate:3441596188 1 udp 659136 10.3.1.1 20622 typ host generation 0 a=candidate:3441596188 1 udp 659136 50.18.X.X 20622 typ srflx generation 0 a=candidate:3441596188 2 udp 659136 10.3.1.1 20623 typ host generation 0 a=candidate:3441596188 2 udp 659136 50.18.X.X 20623 typ srflx generation 0 For instance, if the RTP port was 20622, the RTCP port would be 20623. We're getting no audio when trying to make Chrome to Android and Android to Android WebRTC calls, and we thought we'd try a different RTCP method as we did have it working with Asterisk but want to use FreeSWITCH instead. I've checked out the latest FreeSWITCH code from the master branch. Not sure if FreeSWITCH can do this yet, but if anyone knows that would be helpful. Thank you! James

3 13

Caller ID Name = Outbound Call
by Roger Castaldo 04 Dec '13

04 Dec '13

Hi everyone, I am having a difficult time with an idea that I thought made sense in my head but apparently isn't working how I think it should and need to know what I am doing wrong. I originate a call using the event socket originate {userid=1234,execute_on_answer='transfer call_user_into_conference_123456 XML public',ignore_early_media=true,origination_caller_id_name='1234',originate_timeout=120,origination_caller_id_number='1234' }user/1000 &park() This should and does execute the extension <extension name="call_user_into_conference"> <condition field="destination_number" expression="^call_user_into_conference_(\\d+)$"> <action application="playback" data="conference/8000/welcome.wav"/> <action application="conference" data="$1@wideband"/> </condition> </extension> when I answer the phone, testing the call, it plays the file then moves me into the conference which is perfect. The problem is I have code monitoring Caller-Caller-ID-Name and that gets moved to Outbound Caller some time between me answering the call and conferencing it, I have tried setting effective_caller_id_name and such but it still changes my caller id name to "Outbound Call". Any advice on how to fix this would be awesome.

2 2

FS codec negotiation...
by rumodigital . 04 Dec '13

04 Dec '13

hello guys! I'm migrating from Asterisk to FS... after a few hours studying and am very pleased with what I have seen. I take the opportunity to ask a question about the codecs: if I have a softphone or hardphone with different codecs no way to communicate? eg. Doug (A) and Bob (B) (A) codecs: G722,PCMU (B) Codecs: GSM,Speex FS list codecs: G722,PCMU,PCMA,GSM,Speex profile: ... <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> ... codecs FS: codec, G.722, mod_spandsp codec, GSM, mod_spandsp codec, Speex, mod_speex Doug dials Bob is possible transcoding by FS? here does not work, I'm missing some detail? Regards to all! Doug

1 0

freeswitch SLA and Yealink phones
by Jason Holden 04 Dec '13

04 Dec '13

I am currently testing a t32g and t46g and am experiencing the following problems. The t32g is set under account / advanced for shared line as broadsoft SCA and under DSC the line1 is set to shared. When I press the line key to place a call I receive the message shared line unavailable. On a polycom phone with shared line I am able to place an outbound call using a shared line can anyone provide some advise on this? As well I have a t46G but when I select account / advanced it does not allow me to change the share type from disabled and I can not figure out why. I've already updated the firmware as for Yealink support's advise. Any help on the above would be much appreciated. Thanks.

2 1

Recording Calls
by Hélder Pinheiro 04 Dec '13

04 Dec '13

Hello all, I just recently got to know FreeSwitch and I am very well surprised with it. I am trying to implement a solution using Freeswitch, where a call is established, and in case of the other end caller digits a specific key/set of keys(DTMF), the call starts being recorded. Do you know if FreeSwitch supports this? and if so, could you give me an overview how to do it? Thank you. Regards, Hélder Pinheiro

3 2