May 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Userspy command
by Stefano Favaro 26 May '20

26 May '20

Hello I'm trying to use userspy command. Is there anyone using this command and working ? It seems not automatic, documentation says: "You will then add to the dialplan an extension where the action is like: <action application="userspy" data="1000(a)192.168.1.1 ${hash(select/${domain_name}-spymap/1000}"/> The two parameters are user@domain to listen to future calls , and an optional uuid of an active call to immediately begin to barge in on. In between active calls, you'll be listening to MOH. Actually it only works for the first call to the userspy extension if the call is already established but it seems not looking for new calls. Thanks. Stefano Favaro R&D +39 345 6636386 +39 0423 733320 stefano.favaro(a)edistar.com Edistar Srl Via Artigianato 1 31050 Vedelago (TV), Italy Parte di Your Voice SpA YourVoice.com | LinkedIn

1 0

Help Troubleshooting Freeswitch Lockup
by Daniel Schreiber 25 May '20

25 May '20

Hi, today I had a freeze of freeswitch on a machine running BigBlueButton (which ship their own freeswitch packages: FreeSWITCH Version 1.10.4-dev+git~20200515T214527Z~e1f63ce5b1~64bit (git e1f63ce 2020-05-15 21:45:27Z 64bit)) I had this kind of lockup 3 times in the last 2 months and I tried to track down the cause of the problem but I failed. BigBlueButton uses the conference module for audio switching. All connections come in via a Websocket. In front of the FreeSwitch websocket there is an nginx reverse proxy which says it cannot connect to the upstream (FreeSwitch) server when the lockup occurs. We have setup conference dialin from our Cisco call manager. During the lockup FreeSwitch does not accept connections from the call manager as well. Today no calls from call manager were involved. In the logs I observed that at some time almost all participants of a conference were thrown out and calls terminated with MEDIA_ERROR hangup cause. Participants came from different autonomous systems, I checked that there was no network interruption. CPU usage on the system was below 20%, memory below 40%. The number of participants during start of lockup was 34 distributed across 8 conferences. During the 40 minutes lockup 4 conferences ended, the others continued to switch. FreeSwitch continued to operate as expected after ca 40 Minutes without any operator interactions. Could you please give some advice on how to find the cause of this problem? Thanks alot, Daniel -- Daniel Schreiber Facharbeitsgruppe Systemsoftware Universitaetsrechenzentrum Technische Universität Chemnitz Straße der Nationen 62 (Raum B303) 09111 Chemnitz Germany Tel: +49 371 531 35444 Fax: +49 371 531 835444

1 0

Freeswitch Stop Responding After 2 Days
by Lloyd Aloysius 25 May '20

25 May '20

Hi All This is happening in same pattern after 2 days.A restart service solve the issues. I tried v1.6 , v1.8, v1.10 same behavior. Can someone help me to find the correct direction. Attached following command output. top -b -H -p `cat /usr/local/freeswitch/run/freeswitch.pid` -n 1 Machine have 42 GB memory. When I look into this output I assume freeswitch use 79g memory. Looks like not enough memory. Am I correct? Even I increase the memory some thing happening after 2 days. I tried physical vs virtual esxi same pattern problem How to solve this problem? Can someone point the correct direction. The following is startup script. Any suggestions? *Start up script* /etc/systemd/system/freeswitch.service ; This file in installations built from Master can be found in ; /usr/src/freeswitch.git/debian ; or ; /usr/src/freeswitch/debian [Unit] Description=freeswitch After=syslog.target network.target local-fs.target mysql.service apache2.service [Service] ; service Type=forking PIDFile=/usr/local/freeswitch/run/freeswitch.pid ;PermissionsStartOnly=true ; blank ExecStart= line flushes the list ExecStart= Environment="USER=freeswitch" Environment="GROUP=freeswitch" ExecStart=/usr/local/freeswitch/bin/freeswitch -u freeswitch -g freeswitch -nc TimeoutSec=45s Restart=always ; exec WorkingDirectory=/usr/local/freeswitch/bin ;User=root ;Group=daemon LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 LimitSTACK=250000 LimitRTPRIO=infinity LimitRTTIME=infinity IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 ;NoNewPrivileges=false [Install] WantedBy=multi-user.target

3 6

freeswitch whatsapp integration?
by sawun 22 May '20

22 May '20

Hello, i am new to FS. Can we use FS to send calls to whatsapp number? Thx -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

2 1

mod_xml_cdr - attended transfer
by Gregor Nanger 21 May '20

21 May '20

Hi! I found some bug in mod_xml_cdr module. Hard to explain, so let's describe scenario: 1. Inbound call to extension 2. extension do attended transfer to external number 3. external number picks up call 4. extension end call 5. inbound leg and external leg are connected This is everything ok. FS does as it should. But when all calls end, I get CDR post to web server only for legs from point 2 upward. I am missing main CDR post, where it is <applog> etc... I get only clone-of flows of main leg, but no CDR of main leg. Theoretically, I should CDR for leg in step 1, for leg in step 2 and for leg in step 5. So, I am missing CDR post for step 1. Interesting is that if in step 4., if external number ends call, extension is connected back to inbound leg and also I get all CDR posts. Hope you understand what I mean. Does anyone handle CDR with attended transfer? BR, Gregor

1 1

freeswitch whatsapp integration?
by sawun 21 May '20

21 May '20

Hello, i am new to FS. Can we use FS to send calls to whatsapp number? Thx -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

2 1

Freeswitch use DTLS v1.0 instead of DTLS v1.2
by François-Xavier Geneste 20 May '20

20 May '20

Hello guys, I'm facing a big trouble for several hours ago and need help.... I'm using Freeswitch v1.10.2 with webRTC successfully installed and running. On the user/webphone side, I'm using Chrome 81.0.4044.138. Incoming and outgoing calls works fine with my webphone stack on my browsers (Firefox, Chrome). No warnings or errors at both sides. But when I do the following scenario with a webphone that can manage several calls at the same time (multi-line feature), it does not work : 1. make a first call routed to a webrtc extension, answer it and keep it connected 2. make a second call routed to the same extension, do not answer and keep the first call connected 3. make a third call routed to the same extension and hold the first line to accept this new call=> when I try to answer this 3rd call, the call is always dropped After digging into logs, and packets captured with wireshark, I found that when the freeswitch try to exchange with the browser to negociate SRTP flow for the 3rd call, it use DTLS v1.0 protocol (instead of v1.2) : Unfortunately, support for DTLS v1.0 seems to have been dropped on my webphone/browser side and the freeswitch fail on last DTLS exchange with this logs : [INFO] switch_rtp.c:3736 Activate RTP/RTCP audio DTLS client [INFO] switch_rtp.c:3903 Changing audio DTLS state from OFF to HANDSHAKE [...] [ERR] switch_rtp.c:3266 audio Handshake failure 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is set) but endpoint requires DTLS v1.2. On freeswitch side, I found only one option linked to the DTLS version (legacyDTLS, as written in logs) which I never set in my config. I checked my open ssl version on the freeswitch server (1.1.1d). The thing that is disturbing to me is that if I hold the first call and answer the second call, it works well. The issue occurs only for the third call and after a missed/refused call while still connected with first call in parallel. Digging into freeswitch source, I found that it seems to use version-flexible DTLS methods of openssl (DTLS_server_method() and DTLS_client_method()) and I cannot see how to quicly and simply always force DTLS v1.2 ? Have any of you ever had this kind of problem or know how to solve it ? Regards, FX

4 5

Correlation between jitter/discarded_packets with choppy audio streams
by Unai Rodriguez 20 May '20

20 May '20

Dear List, We're experiencing choppy/distorted audio streams at times and have been looking into understanding the definition and units of measure of these values reported on the main FS log (i.e. /var/log/freeswitch/freeswitch.log) - r = received - l = lost - j = jitter - p = playout_delay - d = discarded_packets - i = ignored_packets The following log lines seemed to correspond with choppy/distorted audio: ======================================================== ... 2020-05-15 10:18:05.202714 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver 10.252.148.75:18800 <- 10.252.148.73:17592 [r:403 l:0 j:651 p:0 d:87 i:0] ... 2020-05-15 10:25:53.622708 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver 10.252.148.75:22660 <- 10.252.148.56:18572 [r:402 l:0 j:143 p:0 d:84 i:0] ... 2020-05-15 10:43:42.642722 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver 10.252.148.75:18940 <- 10.252.148.72:20800 [r:356 l:0 j:838 p:0 d:79 i:0] ... 2020-05-15 11:00:52.102707 [INFO] mpf_rtp_stream.c:541 () Close RTP Receiver 10.252.148.75:20952 <- 10.252.148.61:16540 [r:582 l:0 j:806 p:0 d:66 i:0] ... ======================================================== At this point we believe that having a (d)iscarded value of more than 60 correlates with choppy calls and most of the time those come with (j)itter value of several hundreds. Would it be possible for someone to point out where we could find the definition and unit of measure of those values (i.e.: r, l, j, p, d, i)? Is there any baseline that could be references to determine which calls are being choppy/distorted based on these numbers? We've been trying to understand FS source code and find documentation to no avail. Thank you so much Unai Rodriguez

3 2

oriiginate_timeout
by David Villasmil 20 May '20

20 May '20

Hello all, I'm trying to timeout an origination, using originate_timeout works fine _if_ i don't get a 183, so i tried combining both originate_timeout and bridge_answer_timout, but this doesn't seem to be working: bgapi expand originate {bridge_answer_timeout=5,originate_timeout=2,execute_on_answer='sched_broadcast +3 playback somefile.wav'}sofia/outbound/1112223456(a)1.2.3.4 If i don't get a 183 before 2 seconds, the call is timed out properly, but if i do, the bridge_answer_timeout is never fired.. Anyone knows what this could be? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 7

FreeSWITCH 1.6 EOL
by Mike Jerris 19 May '20

19 May '20

FreeSWITCH 1.6 has been EOL for quite some time. Please note that we will be removing some old packaging repositories for 1.6 on June 1st. If you have not upgraded to a newer version, please note that 1.6 has many severe security issues and we would strongly encourage you to upgrade ASAP. Thanks Mike

1 0