- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Timing issues with Verto, SIP client works fine
by Support from NetworkedAudio LLC 22 Jun '20

22 Jun '20

Hi all, We're using VERTO in an intranet application. If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even"). This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS. Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link. Using exactly the same computers and headsets but using MicroSIP, everything is perfect. Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system. Where should I look now? Have tried 1.10 but no change.

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Missing mod_voicemail_callpage
by António Silva 22 Jun '20

22 Jun '20

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 <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

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Any experiences with mod_portaudio on raspberry pi 4 with buster lite?
by saiedt 20 Jun '20

20 Jun '20

Hi, portaudio is not finding any audio devices: 1. Pi-4 with Raspbian buster lite und latest related FreeSwitch distribution, apt-upgraded and updated just yesterday 2. aplay is working both with the audio jack and with bluetooth (although not simultaneously: as soon as I have a bluetooth audio connection, the audio jack stops to work, and I cannot switch back to it with "amixer cset numid=3 1", but aplay can continue to use the bluetooth connection again after I issue "amixer cset numid=3 65536") 3. it seems that I do not have any permission issue as both users "pi" and "freeswitch" are in the group "audio" which has rw-access to all related devices under /dev/snd 4. I have already installed alsaoss, libasound2 and libasound2-dev 5. The output of "pa rescan" in fs_cli is the following: 2020-06-19 06:55:18.803478 [INFO] mod_portaudio.c:3186 Looking for new devices. 2020-06-19 06:55:18.803478 [INFO] mod_portaudio.c:2102 PortAudio version number = 1246720 PortAudio version text = 'PortAudio V19.6.0-devel, revision 396fe4b6699ae929d3a685b3ef8a7e97396139a4' 2020-06-19 06:55:18.943433 [DEBUG] mod_portaudio.c:1773 global indev [-1] 2020-06-19 06:55:18.943433 [ERR] mod_portaudio.c:1777 Cannot find an input device 2020-06-19 06:55:18.943433 [DEBUG] mod_portaudio.c:1786 global outdev [-1] 2020-06-19 06:55:18.943433 [ERR] mod_portaudio.c:1790 Cannot find an output device 2020-06-19 06:55:18.943433 [INFO] mod_portaudio.c:2125 Number of devices = 0 Any clue, what could be wrong on my side? Thanks! -- Saied -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Re: [Freeswitch-users] nested dialplan
by David P 19 Jun '20

19 Jun '20

Note that you have a typo 'date' instead of 'data' in some of your attributes.

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nested dialplan
by David Villasmil 19 Jun '20

19 Jun '20

Hello all, why does this: <extension name="catch_simple_message_from_originate"> <condition field="destination_number" expression="^(simple_message)$" require-nested="false"> <action application="ring_ready"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="${audio}"/> <condition field="${record_voicemail}" expression="^true$"> <action application="log" data="CRIT I will be recording!"/> <action application="record_session" data="tmp/${record_filename}.mp3"/> </condition> <action application="hangup" date="NORMAL_CLEARING"/> </condition> </extension> Why when executed is moved around? The hangup should execute AFTER the inner condition Dialplan: sofia/external/1234567890(a)sip.mydomain.com Action sleep(1000) Dialplan: sofia/external/1234567890(a)sip.mydomain.com Action playback(${cdrBucket}/${audio1}) Dialplan: sofia/external/1234567890(a)sip.mydomain.com Action hangup() |--- Dialplan: Processing recursive conditions level:1 [catch_simple_message_from_originate_recur_1] require-nested=FALSE |--- Dialplan: sofia/external/1234567890(a)sip.mydomain.com Regex (PASS) [catch_simple_message_from_originate_recur_1] ${record_voicemail}(true) =~ /^true$/ break=on-false |--- Dialplan: sofia/external/1234567890(a)sip.mydomain.com Action log(CRIT I will be recording!) |--- Dialplan: sofia/external/1234567890(a)sip.mydomain.com Action record_session(tmp/${record_filename}.mp3) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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