February 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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Google Chrome Version 81.
by MARAND, Remi 04 Mar '20

04 Mar '20

We use Freeswitch in the case of a SIP to WebRTC application and it is a solid and reliable solution. When we test calls with the sipjs.js library (V0.15.10) and Freeswitch (FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) + Debian 9, with the beta version of Chrome, Version 81.0.4044.17 (Official Build) beta (64-bit), we observe a notification on the javascript console : "[Deprecation] Your partner is negotiating an obsolete (D)TLS version. Support for this will be removed in M81, around March 2020. Please check with your partner to have this fixed. » The Wireshark trace shows that the DTLS-SRTP exchange is done with DTLSv1.0. What would be the version update to do on the Freeswitch side (or Debian) to avoid this deprecation ? Is there a configuration to do in our current version? Best regards. Rémi Marand Product Owner | Marketing & Product Odigo France | Issy les Moulineaux Tel.: +33 1 46 84 12 77 / +33 6 87 72 53 25. www.odigo.com<http://www.odigo.com/> This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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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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JsSip Client receiving "488 Not Acceptable Here" when trying to make a call
by Mahmood Alkhalil 04 Mar '20

04 Mar '20

Hi Everyone, I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here. The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found. I'v attached the call trace, appreciate any help. Thanks, Mahmood Alkhalil.

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Help for Google Chrome version 81 & FS.
by MARAND, Remi 03 Mar '20

03 Mar '20

We use Freeswitch in the case of a SIP to WebRTC application and it is a solid and reliable solution. When we test calls with the sipjs.js library (V0.15.10) and Freeswitch (FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) + Debian 9, with the beta version of Chrome, Version 81.0.4044.17 (Official Build) beta (64-bit), we observe a notification on the javascript console : "[Deprecation] Your partner is negotiating an obsolete (D)TLS version. Support for this will be removed in M81, around March 2020. Please check with your partner to have this fixed. » The Wireshark trace shows that the DTLS-SRTP exchange is done with DTLSv1.0. What would be the version update to do on the Freeswitch side (or Debian) to avoid this deprecation ? Is there a configuration to do in our current version? Best regards. Rémi Marand Odigo France | Issy les Moulineaux Tel.: +33 1 46 84 12 77 / +33 6 87 72 53 25. www.odigo.com<http://www.odigo.com/> This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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Freeswitch 32bits
by John Tuxies 28 Feb '20

28 Feb '20

Trying to install freeswitch in a debian 10 32bits (ALIX 2d13) machine from source and i am getting the following error: *root@alix2d13: ~ $ apt-get build-dep freeswitchReading package lists... DoneReading package lists... DoneBuilding dependency treeReading state information... DoneSome packages could not be installed. This may mean that you haverequested an impossible situation or if you are using the unstabledistribution that some required packages have not yet been createdor been moved out of Incoming.The following information may help to resolve the situation:The following packages have unmet dependencies: builddeps:freeswitch : Depends: libks but it is not installable Depends: signalwire-client-c but it is not installable Depends: libbroadvoice-dev but it is not installable Depends: libsilk-dev but it is not installable Depends: libv8-6.1-dev but it is not installableE: Unable to correct problems, you have held broken packages.root@alix2d13: ~ $* No idea how to overcome this issue.

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