December 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

asr_handle codec = (null)
by Bernd Krueger-Knauber 12 May '20

12 May '20

Hi, I'm testing unimrcp but get empty rtp packets from fs to unimrcp server when I want detect speech. The support from unimrcp told me that with my unimrcp setup is everything fine. He pointed me to the following log entry: [INFO] mod_unimrcp.c:3134 asr_handle: name = unimrcp, codec = (null), rate = 8000, grammar = (null), param = uni2 codec = (null) is wrong. But I have no idea what I can do. Everything else is working in this fs installation. Also speech generation via unimrcp is working. I use FreeSwitch 1.8.7. Any hints?

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mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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FYI - new Amazon Web Services TTS option
by Nathan Neulinger 30 Apr '20

30 Apr '20

https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-l… The quality blows away everything I've tried out previously... Highly relevant option for voice prompts. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Problems with TLS after upgrading to Buster
by Walter Behrend 06 Apr '20

06 Apr '20

Hello there, hope someone else also had the problem - and found a solution for it. My "internal" profile has TLS enabled with tlsv1, 1.1 and 1.2 - this worked like a charm on stretch. I'm using the freeswitch-repos. I upgraded to buster and here my problems started. Seems the gentls_cert only creates SHA1 (CA)Certificates - so freeswitch started with openssl error messages "md too weak". Tried at first to bypass this error by setting the tls_ciphers to "DEFAULT:@SECLEVEL=0" but this error still occured. So as a consequence, I modified the gentls_cert script and replaced everywhere the parameter -sha1 with -sha256. This error disappeared now, but the next one is coming up. It seems it does not matter which value I set for "tls_version" - in every case, my TLS enabled port only accepts TLS 1.3 connections. I have the problem that we're also using older phones which only support TLS 1.0. Error message is: tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol I tried with openssl s_client and the parameters -tls1 -tls1_1 and so on - it really only worked for -tls1_3 Any idea about this? setting tls_version to tlsv1,tlsv1.1,tlsv1.2 does not change anything. Also setting the value just to tlsv1 does not help, I verified this with the phones AND with openssl s_client. Still only TLS 1.3 gives results here. Thanks in advance... BR Walter

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dynamic music on hold for mod_conference
by Lappe, Adam 19 Mar '20

19 Mar '20

Hi all, i am trying to dynamically set the mod-sound variable in mod_conference, so I can play different music on hold to my conference rooms without having to create multiple profiles. Is there a way to set my moh-sound path while I create the conference? Or can I change the moh-sound file via the event socket in a existing conference? I think this would also lead to success. Thanks in advance, Adam

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New Debian FS vanilla install fails
by Sean Devoy 04 Mar '20

04 Mar '20

HI all, I have a new Debian 8.1 (32 bit) server for testing. I first did an apt-get install curl then per the instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Debian+7 I did this: echo "deb http://files.freeswitch.org/repo/deb/debian/ `lsb_release -sc` main" >> /etc/apt/sources.list.d/freeswitch.list curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - apt-get update apt-get install freeswitch-meta-vanilla It runs for less than a minute and dies at: Setting up freeswitch-sysvinit (1.4.19~10~1-1~jessie+1) ... Job for freeswitch.service failed. See 'systemctl status freeswitch.service' and 'journalctl -xn' for details. invoke-rc.d: initscript freeswitch, action "start" failed. dpkg: error processing package freeswitch-sysvinit (--configure): subprocess installed post-installation script returned error exit status 1 Setting up freeswitch-systemd (1.4.19~10~1-1~jessie+1) ... dpkg: dependency problems prevent configuration of freeswitch-init: freeswitch-init depends on freeswitch-sysvinit (= 1.4.19~10~1-1~jessie+1); however: Package freeswitch-sysvinit is not configured yet. dpkg: error processing package freeswitch-init (--configure): dependency problems - leaving unconfigured and ends shortly with: Errors were encountered while processing: freeswitch-sysvinit freeswitch-init E: Sub-process /usr/bin/dpkg returned an error code (1) I ran the cmds it says: systemctl status -l freeswitch.service * freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; disabled) Active: failed (Result: start-limit) since Wed 2015-06-17 17:11:15 EDT; 18min ago Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service: control process exited, code=exited status=1 Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. journalctl -xn -- Logs begin at Wed 2015-06-17 16:59:45 EDT, end at Wed 2015-06-17 17:17:01 EDT. -- Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service: control process exited, code=exited status=1 Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. -- Subject: Unit freeswitch.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit freeswitch.service has failed. -- -- The result is failed. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. -- Subject: Unit freeswitch.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit freeswitch.service has failed. -- -- The result is failed. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:54 fstest.devoy.us kernel: perf interrupt took too long (10077 > 10000), lowering kernel.perf_event_max_sample_rate to 12500 Jun 17 17:17:01 fstest.devoy.us CRON[12430]: pam_unix(cron:session): session opened for user root by (uid=0) Jun 17 17:17:01 fstest.devoy.us CRON[12431]: (root) CMD ( cd / && run-parts --report /etc/cron.hourly) Jun 17 17:17:01 fstest.devoy.us CRON[12430]: pam_unix(cron:session): session closed for user root Did I misunderstand that Debian 8 was OK, in fact required at 1.6.n? What am I missing? Sean

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Refer and Cannot Blind Transfer 1 Legged calls
by David Villasmil 07 Feb '20

07 Feb '20

Hello all, I'm trying to create a new INVITE using REFER from kamailio. Sending FS the first leg (which is answered in the dialplan and sent to a python script.) And then sending a REFER to FS to call another endpoint. I've tried and i'm always getting *Cannot Blind Transfer 1 Legged calls* and getting a NOTIFY with 404 Not Found Should this work? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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No audio for conference call
by Md,Mehedi Hasan Kabir(Tanim) 23 Jan '20

23 Jan '20

Hi I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this? Audio is fine when I call other users directly. Regards Tanim

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uuid_simplify after intercept/bridge
by Mitchell Langs 07 Jan '20

07 Jan '20

Hi, I need to originate 2 calls and later bridge them. Then Freeswitch needs to be removed from the call path. uuid_simplify accomplishes this. But where do I put this command so that it is executed once the two calls are bridged? As far as I understand the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge" command until the bridge is terminated. I have also tried to set sip_auto_simplify=true but it had no effect. Here's what I am executing: originate {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620 testing XML public <extension name="testing"> <condition field="destination_number" expression="^testing$"> <action application="set" data="api_result=${bgapi originate {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/> ... <action application="playback" data="$${hold_music}"/> </condition> </extension> Kind Regards

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