June 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Compilation of Vanilla FreeSWITCH on Centos 7
by Kumar, Puneet (Puneet) 29 Jun '20

29 Jun '20

Hello, I followed steps given at link https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 to compile FreeSWITCH v1.10.3 code. But getting following errors on make : CC libavmod_la-mod_av.lo CC libavmod_la-avformat.lo In file included from /usr/include/ffmpeg/libavutil/imgutils.h:31:0, from avformat.c:39: /usr/include/ffmpeg/libavutil/pixdesc.h:34:5: error: type of bit-field 'plane' is a GCC extension [-Werror=pedantic] uint16_t plane : 2; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:40:5: error: type of bit-field 'step_minus1' is a GCC extension [-Werror=pedantic] uint16_t step_minus1 : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:46:5: error: type of bit-field 'offset_plus1' is a GCC extension [-Werror=pedantic] uint16_t offset_plus1 : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:52:5: error: type of bit-field 'shift' is a GCC extension [-Werror=pedantic] uint16_t shift : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:57:5: error: type of bit-field 'depth_minus1' is a GCC extension [-Werror=pedantic] uint16_t depth_minus1 : 4; ^ avformat.c:44:38: fatal error: libavresample/avresample.h: No such file or directory #include <libavresample/avresample.h> ^ cc1: all warnings being treated as errors compilation terminated. Am I missing something? Thanks, Puneet

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Drop A leg and originate a B-leg
by David Villasmil 28 Jun '20

28 Jun '20

Hello all, So right now I'm generating A simple INVITE, let's say from FS-G. On receiving the INVITE, FS launches a python script which should reject the incoming INVITE from G and start a new call B-leg. I already do this, but the FS-G leg keeps going and I would like to actually reject and hangup the FS-G leg. is this possible? Basically hangup the incoming call and keep the script running to create a new leg. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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OSError when using Python ESL + socketserver
by René Weiss 26 Jun '20

26 Jun '20

Hi, When I'm using the ESL library (python-ESL==1.4.18) for Python (3.7.7) together with socketserver I always get an OSError when the connection is closing. Here some sample code (mostly the example from the wiki ported to Python 3): ----------- import socketserver import ESL class ESLReqHandler(socketserver.BaseRequestHandler): def setup(self): print(self.client_address, 'connected!') fd = self.request.fileno() print(fd) con = ESL.ESLconnection(fd) print('Connected: ', con.connected()) if con.connected(): info = con.getInfo() uuid = info.getHeader("unique-id") print(uuid) con.execute("answer", "", uuid) con.execute("sleep", "2000", uuid) server = socketserver.ThreadingTCPServer(('', 8040), ESLReqHandler) server.serve_forever() ----------- And this is the error message I get: ----------- Exception in thread Thread-1: Traceback (most recent call last): File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/threading.py", line 926, in _bootstrap_inner self.run() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/threading.py", line 870, in run self._target(*self._args, **self._kwargs) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 654, in process_request_thread self.shutdown_request(request) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 509, in shutdown_request self.close_request(request) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 513, in close_request request.close() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socket.py", line 420, in close self._real_close() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socket.py", line 414, in _real_close _ss.close(self) OSError: [Errno 9] Bad file descriptor ----------- The problem starts to appear as soon as the command "ESL.ESLconnection(fd)" is included in the setup function. Any ideas? Thanks René Weiss

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Noise on line, does not show up in recordings
by Sam van Herwaarden 26 Jun '20

26 Jun '20

Hi all, I frequently get noise during calls with my FreeSWITCH deployment. It sounds like a brief burst of distortion - typically less than a second. I've had it happen both when using playback with WAV (different formats - I found recommendations suggesting 8000Hz, 128 kb/s mono and it shows up with that format as well) and text-to-speech with flite. It happens at unpredictable moments - it's not the same from one call to the next. I've added a record_session statement to investigate in more detail, but the noise does not show up in the recording. Does anyone have advice on what might be causing this or how I could further debug this? Kind regards, Sam

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http_cache upload failure
by David Villasmil 25 Jun '20

25 Jun '20

Hello all, I'm doing: <action application="record" data="http_cache:// https://mybucket.s3.amazonaws.com/file.mp3 30 3 2"/> and i'm getting 2020-06-25 02:14:18.251194 [NOTICE] sofia.c:1079 Hangup sofia/external/ vozelia-prod-1@37.139.121.39:5080 [CS_EXECUTE] [NORMAL_CLEARING] 2020-06-25 02:14:18.251194 [DEBUG] mod_http_cache.c:342 opening /var/cache/freeswitch/b6/349a08-37ac-409d-b607-0c0b72369d02.mp3 for upload to https://mybucket.s3.amazonaws.com/file.mp3 2020-06-25 02:14:18.291196 [ERR] mod_http_cache.c:431 Received HTTP error 0 trying to save /var/cache/freeswitch/b6/349a08-37ac-409d-b607-0c0b72369d02.mp3 to https://mybucket.s3.amazonaws.com/file.mp3 but i can upload from the CLI curl -X PUT -T "ef8d02-5738-4c9b-886f-934e8a5fc051.mp3" https://mybucket.s3.amazonaws.com/file.mp3 Works perfectly... Any ideas? Many thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Incoming call audio issue
by Roberto Scarpone 25 Jun '20

25 Jun '20

Hi, I have a problem when phones receive a call, for a time varying between 1-3 seconds, they hear nothing (at the beginning of the call). The problem does not always occur, but with a frequency of 2/3 time every 10 calls. After a few seconds the conversation presents no further problems. Listening to the captured audio the call doesn't seems to have any problem, the audio is complete, but when they answer the call the problem is there and they can't hear the first 1-3 seconds of the conversation. I have a FreeSWITCH 1.6.20 on Debian 8, installed from repo. Has anyone already encountered this problem or would you know how to help me solve it? Best regards, Roberto

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session:setVariable without logging it
by Michael Smeltzer 24 Jun '20

24 Jun '20

Hi all, I'm trying to use session:setVariable in my script and noticed that when I do this, it shows up in the CLI and freeswitch.log files as DEBUG. 2020-06-23 23:22:23.369212 [DEBUG] switch_cpp.cpp:755 CoreSession::setVariable(variable_name, variable_value ) Is there a way to prevent this from getting logged? I know I can turn off DEBUG level logs but there is other DEBUG-level information that is useful. Alternatively, is there another way to pass a variable to a python script besides what is documented here that requires you to set/get session variables? I tried various implementations of passing it as an argument but was not successful in figuring out the correct format. https://freeswitch.org/confluence/display/FREESWITCH/mod_python#mod_python-… Thanks, Michael

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TLS setup
by ramelcom 24 Jun '20

24 Jun '20

Hi, I am trying to setup TLS in FreeSWITCH. In my usecase, FS acts as a server and another SIP B2BUA acts as the client. I was able to successfully setup the TLS if I generate the certificates at FS and configure with those. However, if I want to use server certificate generated and signed by 3rd party, it doesn't work. I added server certificate as agent.pem and root CA as cacert.pem and cafile.pem. Also, I am configuring the root CA at SIP B2BUA side. When SIP B2BUA sends Client Hello, FS rejects with 'Handshake failed' error. Can you please help on this? Thanks Ramesh -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Support for getDisplayMedia API in Verto JS
by Jonathan Hunter 23 Jun '20

23 Jun '20

Hi Guys, I just wondered if anyone has got this working with verto and freeswitch 1.10 ? I can grab the screen I want to share, and it makes a call to freeswitch using the -screen dialplan but it doesnt seem to work correctly as other participants cant see it, just the conference member sharing the screen. Any tips where I might be going wrong? Previously I have used a plugin but trying to move away from that now. Many thanks! Jon

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Re: [Freeswitch-users] Missing mod_voicemail_callpage
by Giovanni Maruzzelli 23 Jun '20

23 Jun '20

maybe it was in "contrib"? https://freeswitch.org/stash/projects/FS/repos/freeswitch-contrib/browse On Mon, Jun 22, 2020 at 3:22 PM António Silva via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "António Silva" <asilva(a)wirelessmundi.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 22 Jun 2020 15:22:05 +0200 > Subject: Missing mod_voicemail_callpage > Hi, > I want to implement a script that calls a user when a new message is left > in the voicemail. > And I found in confluence reference to an interesting module voicemail > callpage that places a call to a destination when new messages are present > in a voicemail box: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail_callpage > > But this module is not present at source code (I couldn’t found ..) > > Is it only available for commercial solutions? > > > Thanks, > António > > > > ---------- 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 Jun 2020 06:22:42 -0700 (PDT) > Subject: [Freeswitch-users] Missing mod_voicemail_callpage > _________________________________________________________________________ > > 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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