January 2020 - 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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Re: [Freeswitch-users] Limit cps per source IP and port
by Mickael Hubert 14 Feb '20

14 Feb '20

Maybe it's very late, but I use this: https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml it's not per IP, but it's easy to change this code Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> a écrit : > > > > ---------- Forwarded message ---------- > From: "António Silva" <asilva(a)wirelessmundi.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 18 Jul 2019 10:52:14 +0200 > Subject: Limit cps per source IP and port > Hi, > > is it possible to limit the number of calls per source ip and port > before the dialplan? > > I know that this can be done using mod_limit in the dialplan, i was > thinking of a limit like the sessions-per-second global limit, this way > less resources are used if some ip/port is flooding our machine. > > Another possibility is using the firewall... but i like this option less > because i wont know of the drop calls. > > Thanks, > > -- > Saludos / Regards / Cumprimentos > António Silva > > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 22 Jul 2019 14:03:23 -0700 (PDT) > Subject: [Freeswitch-users] Limit cps per source IP and port > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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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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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Calls repeat at 80 seconds?
by Avi Marcus 02 Feb '20

02 Feb '20

FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) Is anyone having an issue with calls suddenly "restarting" at about 80 seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now across various destinations using different carriers. It's not on every call, just some. I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit (-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue. Thanks, -Avi Marcus

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