- FreeSWITCH-users - FreeSWITCH Mailing Lists

Detect Silence question
by Igor Olhovskiy 01 Sep '18

01 Sep '18

Hi! Is there any way to make silence detection on start of the call? Like answer a call, playback a ringtone back to legA, and meanwhile try to understand, if legB is totally silent. If it is silent for, let’s say 5 sec - hangup the call. One of ways to fight spam machine calls (yes, sometimes it looks like DoS attack) Right now only thing I can imagine - record legA to file and than analyze recording for silence. Regards, Igor

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mod_tts_commandline hangs in call flow
by Lee Vidor 31 Aug '18

31 Aug '18

Hi I'm currently using mod_tts_commandline with a python script to AWS text to speech API. The flow works fine during a single call. Event socket API -> Freeswitch dialplan -> mod_tts_commandline -> Python script -> AWS API -> Voice file -> Freeswitch say However when there a number of concurrent calls (even just 10 to 15) the request to execute the python script hangs until all the calls are at the same requesting state. As if mod_tts_commandline is waiting for something to happen. Dial plan: <extension name="tts_google"> <condition field="destination_number" expression="^11111$"> <action application="set" data="call_timeout=60" /> <action application="answer"/> <action application="detect_silence" data="100 30 25000"/> <action application="speak" data="tts_commandline|${lang}|${text}" /> <action application="hangup"/> </condition> </extension> tts_commandline.conf: <param name="command" value="/usr/bin/python /usr/local/freeswitch/scripts/synthesize_text.py --language=${voice} --filename=${file} --text=${text}"/> </settings> Execution of python script in freeswitch logs: 2018-08-30 11:19:57.641892 [ERR] switch_ivr_play_say.c:2385 switch_ivr_detect_silence: SILENCE DETECTED 2018-08-30 11:19:57.641892 [DEBUG] switch_core_codec.c:248 sofia/external/33123456789 Restore previous codec PCMU:0. EXECUTE sofia/external/33123456789 speak(tts_commandline|en-US|Hi, This is a test message) 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3025 OPEN TTS tts_commandline 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3035 Raw Codec Activated 2018-08-30 11:19:57.641892 [DEBUG] mod_tts_commandline.c:160 Executing: /usr/bin/python /usr/local/freeswitch/scripts/synthesize_text.py --language='en-US' --filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav' --text='Hi, This is a test message' "Simultaneous calls requesting mod_tts_commandline" Log from python script: (notice it's called with a 15 second delay) ---> *What is causing this delay?* 2018-08-30 11:20:12,488 DEBUG Called script 2018-08-30 11:20:12,490 DEBUG Arguments Namespace(filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav', language='en-US', ssml=None, text='Hi, This is a test message') 2018-08-30 11:20:12,490 DEBUG synthesize_text requested The python script executes fine and returns to freeswitch: 2018-08-30 11:20:14.541919 [DEBUG] switch_core_file.c:342 File /tmp/c61d2034-c1b1-4f42-932f-350484553b5a.tmp.wav sample rate 24000 doesn't match requested rate 8000 2018-08-30 11:20:14.541919 [DEBUG] switch_ivr_play_say.c:2729 Speaking text: Hi, This is a test message Any help or further debugging steps would be appreciated. Thanks Lee

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mod_flite fails due to undefined symbol: cmu_lex_init
by kaiduan xie 30 Aug '18

30 Aug '18

Hi all, I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so**/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** Any idea to solve this issue? Many thanks, /Kaiduan

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Trickle ICE
by Mirko Brankovic 30 Aug '18

30 Aug '18

Hi, I read a mailing thread from year or two back, about mod_verto and mod_sofia sharing a same media stack, and that implementing Trickle ICE would mean quite some work. I'm wondering if anyone started with this work or if it is already there? I see that right now SDP without any candidate is dropped by the media.c :) Thanks, -- Regards, Mirko

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Re: [Freeswitch-users] wait_for_silence not working through external gateway
by Lee Vidor 30 Aug '18

30 Aug '18

After much trial and error issue is related to firewall settings. Opening ports 16384 - 32768 enabled silence detection to work as expected. On Thu, 23 Aug 2018 at 17:02, vidor via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: vidor <leevidor89(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 23 Aug 2018 08:17:35 -0700 (MST) > Subject: wait_for_silence not working through external gateway > I'm attempting to detect when there is silence on the call in order to play > audio file. If I dial to an internal extension the wait_for_silence and > detect_silence commands are working fine. > > However if dial out through an external gateway to PSTN I don't get > expected > results. Silence is always detected, even if there is voice speaking. > I can detect DTMF commands fine... > > Is there any known issue or parameter required on the gateway in order to > detect the audio / silence. > > Thanks in advance! > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: vidor via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 23 Aug 2018 09:02:48 -0700 (PDT) > Subject: [Freeswitch-users] wait_for_silence not working through external > gateway > _________________________________________________________________________ > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com

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1.8 Packages
by Darren Williams 29 Aug '18

29 Aug '18

Looking great! Thanks guys! Just one problem I see and that is postgres doesn't appear to be compiled in core anymore.

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Re: [Freeswitch-users] JIRA FS-9657 - Inbound HEP SIP capture for TLS transport
by Tom Parrott 29 Aug '18

29 Aug '18

Hi Sergey, Can you explain what you mean? Did you mean you would like RTCP stats added? Thanks Tom On 2018-08-29 13:54, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: JIRA FS-9657 - Inbound HEP SIP capture for TLS transport > (Sergey Safarov) > 2. Re: rtp_has_crypto (Social Boh) > 3. Attended Transfer to Call Center Queue when agent answers at > the same time the transfer is completed (Shaun Stokes) > 4. Re: Attended Transfer to Call Center Queue when agent answers > at the same time the transfer is completed (Shaun Stokes) > _______________________________________________ > 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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rtp_has_crypto
by Social Boh 29 Aug '18

29 Aug '18

Hello list, I'm testing this dialplan Block: <extension name="hold_music"> <condition field="destination_number" expression="^9664$"/> <condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <anti-action application="set" data="zrtp_secure_media=true"/> <anti-action application="answer"/> <anti-action application="playback" data="silence_stream://2000"/> <anti-action application="execute_extension" data="is_zrtp_secure XML features"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> using Bria Softphone; his SDP look like: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BtJ/e81Ed/6DMngVpHxmi+F5WHZQt89R9V1PYv27 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:BtJ/e81Ed/6DMngVpHxmi+F5WHZQt89R9V1PYv27 Why don't match with: ^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$ Dialplan: sofia/internal/1003(a)144.202.15.148 Regex (FAIL) [hold_music] ${rtp_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false Thank you Regards -- --- I'm SoCIaL, MayBe

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REFER does not hit the dialplan on attended transfer
by Peter Steinbach 29 Aug '18

29 Aug '18

I try to find out a solution for the following: * A calls B (Dialplan A to B is executed with A as external number and B as local phone) * B is initiating an attended transfer to C (local phone), Dialplan B to C is executed * B is then executing the attended transfer to C, B sends a REFER to Freeswitch with C in the Refer header) * Freeswitch detects attended transfer, does not trigger a dialplan and redirects the call directly to user/C * Problem: C still sees B as caller_id in his phone display instead of A, dependent on the phone Type (Snom or Yealink) So I am wondering, if I could get the REFER to trigger a dialplan, so I can control the displyed numbers? I searched a bit and found the variables "force_transfer_dialplan" an "force_transfer_context". Looking into the code in switch_ivr.c, I can see that the variables are preset with "XML", resp. "default". Question: Can I use these variables to hit the dialplan on REFER. And if yes, how can I do this? -- With kind regards Peter Steinbach

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JIRA FS-9657 - Inbound HEP SIP capture for TLS transport
by Tom Parrott 28 Aug '18

28 Aug '18

Hi, I've found that Freeswitch does not allow capturing of SIP packets (using the HEP capture-server feature), when the SIP connection is over encrypted TLS transport. Outbound packets are correctly send to the HEP capture-server, even when over an encrypted TLS transport. This has previously been logged, and closed by someone else on this JIRA case. https://freeswitch.org/jira/browse/FS-9657 However I have looked into it and submitted a 3-line patch to get this working properly. I've posted my notes on the JIRA case and emailed the assignee directly, however have not gotten a response yet. My concern is that the JIRA case is already marked as closed and may be forgotten if its not re-opened. Please can this be looked into. Thanks Tom

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