June 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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