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Re: [Freeswitch-users] Can't reduce rtmp from 1280x720
by David P 19 Jun '20

19 Jun '20

Seven Du said: > the codec doesn't change resolution, the source does. so if you record a conference to send rtmp, change the conference canvas size. or you can try explicitly set the size e.g. record {vw=320,vh=240}rtmp://url... attach a log if it still not working. We initially tried to set the rtmp resolution via "video-canvas-size" in conference.conf.xml... Dialplan snippet: <action application="record_session" data="{modname=mod_av}rtmp://${ipOfPeerInCluster}:5150/viewOfUser/${idOfClientSession}"/> <action application="conference_set_auto_outcall" data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p']sofia/gateway/asterisk/${example_destination}"/> <action application="conference" data="${example_destination}@example_conference+flags{endconf|mintwo}"/> conference.conf.xml snippet: <profile name="example_conference"> <param name="domain" value="$${domain}"/> <param name="rate" value="8000"/> <param name="channels" value="2"/> <param name="interval" value="20"/> <param name="energy-level" value="0"/> <param name="caller-controls" value="none"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/> <param name="video-mode" value="mux"/> <param name="video-layout-name" value="1x1"/> <param name="video-canvas-size" value="320x240"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-fps" value="15"/> <param name="max-members" value="2"/>  <param name="endconf-grace-time" value="5"/> </profile> ...and then we tried the av.conf.xml change that I already mentioned. I'm not sure I need {modname=mod_av} in front of my rtmp url, but in case I do, I adopted your suggestion in this way: New change to dialplan snippet: <action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfPeerInCluster}:5150/viewOfUser/${idOfClientSession}"/> ...And it worked! I wonder why this works and the conference.conf.xml setting doesn't. Thanks!

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Can't reduce rtmp from 1280x720
by David P 18 Jun '20

18 Jun '20

On both FSv1.8 and now on FSv1.10.3 (upgraded via packages), rtmp sent by our dialplan is always received in resolution 1280x720 even though we've tried several different ways of reducing it to 320x240. Verto's logging of our gUM constraints verifies that our max resolution for video from the browser to FS is 640x480. In av.conf.xml, we've tried uncommenting and editing this way: <profile name="H264"> <param name="profile" value="baseline"/> <param name="level" value="2"/> FS logging confirms that the bug is using the H264 codec. Is there anything else to try? Cheers, David

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onWSLogin Verto call back when dont want to register just connect.
by Jonathan Hunter 17 Jun '20

17 Jun '20

Hi Guys, I am testing with connecting over Websockets to FreeSWITCH using verto, and I dont want to register with FreeSWITCH, so I have set this in verto.conf; <param name="blind-reg" value="true"/> I then can see that my browser connects with the server when looking at logs and the verto status whos conn_no_reg. "jsonrpc": "2.0", "id": 5, "result": { "message": "logged in", "sessid": "a3109700-5036-68b8-3cbf-60a8628fe452" } default-v4::@ client 172.31.41.66:57212 CONN_NO_REG (WS) However I do not see the callback function onWSLogin triggered, is a requirement for this to be called the end user registering as well as connecting? When I disconnect on WSClose callback works fine, as do events when I make a conference call, its literally just when I connect I dont see onWSLogin events. Is there something I am missing to allow clients to connect without registration? Please note this is a secure environment hence why registration not necessary, running version 1.10.2 Many thanks Jon

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Video Hold for mod_callcenter
by Nathan Stratton 16 Jun '20

16 Jun '20

I know you can do music on hold with mod_callcenter, but is it possible to play a .mp4 with audio and video for users while on hold for an agent? ><> nathan stratton

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Problem listening and sending ivr options
by Jose Zamora 16 Jun '20

16 Jun '20

Hello I have installed freeswitch version 1.10.0 from the source codes. But when connecting the softphones I experienced the following problems. Zoiper (ext 1000) = When calling the ivr I do not hear anything and when pressing the options do not dial DTMF signals to the control panel. Phonerlite (ext 1001) = I hear perfectly the IVR, but when pressing the options not connected the DTMF signals to the control panel. Anyone had this configuration problem? thanks.

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Pre-Built Virtual Machine - Verto not working
by Oni Kuma 16 Jun '20

16 Jun '20

Hello, I am just starting this journey to learn Freeswitch. I've downloaded the prebuilt virtual machine from https://freeswitch.com/index.php/fs-virtual-machine/ and followed the directions accordingly. When i connect to test Verto, it does not work at all. I've tried with Firefox - I accept the cert, enter the info, it selects the audio mic and camera, and then just says "waiting for reconnect". I get nothing. I've tried with Chrome - I accept the cert, enter the info, and it gets me to the next step where it allows me to select the mic and camera ( i even see the camera light on), but I am unable to select the "save" button. I do not even see my own image from the camera. I've followed the instructions exactly, and have even followed how it is performed in the youtube video. Can someone point me in the right direction?

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calls to some registered webrtc endpoints results in NO_USER_RESPONSE
by Babak Yakhchali 16 Jun '20

16 Jun '20

Hi I'm using webrtc with chrome and sipjs as client on freeswitch 1.10. Calling to some endpoints results in NO_USER_RESPONSE. I've not access to client systems and it makes debugging the problem harder. this is the freeswitch log for one of those calls: https://pastebin.freeswitch.org/view/f9a09df4#L7 Does NO_USER_RESPONSE mean that invite is not reached to the client browser or it is reached but no reply is received? can it be related to network problems or NAT? any help would be appreciated thanks

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Re: [Freeswitch-users] playback to soundcard TML instead ringfile with portaudio
by Tom Lynn 16 Jun '20

16 Jun '20

have you considered recording it to a wave file? On Mon, Jun 1, 2020 at 7:18 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 1 Jun 2020 16:17:29 +0200 > Subject: playback to soundcard TML instead ringfile with portaudio > Hi people, > > I am using on my linux system a lot portaudio to place and make phone > calls. > > Inside the portaudio.conf I am using: > <param name="ring-file" value="/opt/freeswitch10/sounds/ring_dd.wav"/> > > that if a phone call comes in it respectively rings on my headset. > > Instead the file, i'd like freeswitch to playback tones via TML: > https://freeswitch.org/confluence/display/FREESWITCH/TGML > > now what I want is to disable the ring-file and make freeswitch in the > dialplan playback to myself (the soundcard) for example this UK ring: > <action application="playback" > data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> > > not the callee should get this ring, freeswitch should playback this > sequence before I execute "pa answer". > > > Any ideas ? > > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 01 Jun 2020 07:18:13 -0700 (PDT) > Subject: [Freeswitch-users] playback to soundcard TML instead ringfile > with portaudio > _________________________________________________________________________ > > 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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2-Stage Dialing using freeswitch
by Vishal Pai 15 Jun '20

15 Jun '20

Hello Everyone Can we use freeswitch for 2-Stage Dialing ? Does anyone tried with the freeswitch? Basically customer dials 0XX980123456 we strip 0XX dial a local number, wait for a dial tone and then send dial 980123456. Thanks Vishal P.

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Re: [Freeswitch-users] ./configure single module?
by David Villasmil 15 Jun '20

15 Jun '20

Thanks Tamer, but I was referring to the confirmation step; not the make and install. :) On Mon, 15 Jun 2020 at 11:00, Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 15 Jun 2020 11:21:20 +0200 > Subject: Re: [Freeswitch-users] ./configure single module? > Not true. > > steps to reproduce: > > 1. get the freeswitch source > 2. ./bootstrap.sh > 3. edit modules.conf (the minimal system) > 4. ./configure --prefix=/opt/freeswitch > 5. make && make install > > now you decide yourself to install at a later point of time further > modules, (my freeswitch source is in /usr/local/src/freeswitch), then > you enter the directory: > > /usr/local/src/freeswitch/src/mod/applications/mod_conference > > and just type make && make install > > Just tested it, works. > > > Have fun! > > > best, Tamer > > > On 2020-06-12 19:15, Mike Jerris wrote: > > No. Configure is one step for all. > > > >> On Jun 12, 2020, at 10:50 AM, David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> > >> Hello > >> > >> Is it possible to issue something to configure a single module? > something like: > >> > >> ./configure mod_conference > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 15 Jun 2020 03:00:13 -0700 (PDT) > Subject: Re: [Freeswitch-users] ./configure single module? > _________________________________________________________________________ > > 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 -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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