- FreeSWITCH-users - FreeSWITCH Mailing Lists

Get/list current event filters
by René Weiss 17 Mar '20

17 Mar '20

Hi, Is it possible to get a list of all currently set event filters (from mod_event_socket)? Thanks, René

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Re: [Freeswitch-users] Freeswitch one way audio problem - microphone on one side not working
by David P 16 Mar '20

16 Mar '20

Some items to check: 1. Does WinShark show the audio is arriving at the destination? 2. If using a browser client, you might be attaching the stream to the video element too late. Attaching should be done via a handler of the ontrack event (which verto does). 3. If using a browser client, and you're missing inbound audio, something might be pausing or muting the video element used by the call. For example, iOS pauses the call video element if any other video element having an audio track plays.

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Send extra custom info (image data) on call to show in target endpoint
by Jose David Jurado Alonso 16 Mar '20

16 Mar '20

Hi, I'm looking for a way to be able to pass extra information (one image data for example) in a call. The purpose is to be able to display a preview image of the caller on the target device (mobile) so that they can see it before taking or rejecting the call. Currently, I have the image saved and send it using a push notification when the call occurs but I would like to skip that step and send it over SIP. In this way, you could send the image on the fly without having to store it on the server. For this, I need to know if there is the possibility of sending the image as data using the SIP protocol, in the SIP INVITE header or using an additional method so that the target endpoint (mobile) can see the image before to take the call. Any ideas on how to do it? José David Jurado Alonso

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how to install virtual communicator on my server
by John Covici 15 Mar '20

15 Mar '20

Hi. I have been trying for a while to install the virtual communicator out of freeswitch/html5/verto on my freeswitch server. However, I get lots of warnings I don't understand and finally when I get a directory named dist and copy it to the website, I get a blank page. I am using fs 10.1.2 which compiles and runs fine on my gentoo system and I tried to execute the commands in the debian8-install.sh, but things are not working at all for me. I have logged all the install files and can send them to anyone who might be able to help me with this. Some of the warnings I get are like this: mnpm mWARN optional SKIPPING OPTIONAL DEPENDENCY: yallist(a)3.1.1 (node_modules/fsevents/node_modules/yallist): and others. I don't know if I have the correct version of nodejs or even if I could get it, since its not available on Debian 10. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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Freeswitch one way audio problem - microphone on one side not working
by Andrej Šimko 13 Mar '20

13 Mar '20

Hello FS community, I have problem with microphone on one side, which is not working when calling. Here is what I use: - FreeSWITCH (Version 1.10.2-release git f7bdd38 2019-12-31 14:01:19Z 64bit) - pa devlist 0;sun50i-a64-audio: - (hw:0,0)(ALSA);2;2;r,i,o 1;allwinner,hdmi: - (hw:1,0)(ALSA);0;2; 2;sysdefault(ALSA);128;128; 3;lavrate(ALSA);128;128; 4;samplerate(ALSA);128;128; 5;speexrate(ALSA);128;128; 6;upmix(ALSA);8;8; 7;vdownmix(ALSA);6;6; 8;default(ALSA);128;128; 9;dmix(ALSA);0;2; - pa looptest is working fine with microphone functional - SIP call is on one subnet without nat and firewall between FS as client and grandstream GXP1610 (nothing special) - sip trace is not showing any error or something, RTP is ok I really dont know what else should by wrong, does anybody have a clue with this situation? Thank you for any advice! Andrej

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Hangup order of the legs
by Tezcan ÇIRAKOĞLU 11 Mar '20

11 Mar '20

Hello everyone, I am tracking freeswitch events via event-socket interface. But i could not get which leg hangs up first during a call by reading header information included in events. Is there a simple way to get which leg hanged up first during a call? Thanks

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DTMF queued [bind_digit_action]
by Gaurang Gohil 11 Mar '20

11 Mar '20

hello all, im trying to add the ability to screen voicemail being left, and allow the caller to drop a voicemail using ivr with a DTMF. im using bind_digit_action to bind the digit and menu-exec-app in ivr, but its not working. bind_digit_action consume the dtmf not passing to ivr application after using the clear_digit_action before going to ivr. here is the dailplan im using, <extension name = XYZ> <condition field="${destination_number}" expression="(.*)"> <action application="set" data="hangup_after_bridge=true"/> <action application="bind_digit_action" data="net,*11,exec:NetC,hold_break,both"/> <action application="digit_action_set_realm" data="demo"/> <action application="clear_digit_action" data="all"/> <action application="answer"/> <action application="ivr" data="abc_ivr"/> </condition> </extension> <menu name="abc_ivr" greet-long="Wlcm" greet-short="Repeat_AutoA" invalid-sound="Invalid_AA" exit-sound="Exit_AA" timeout="10000" max-timeouts="5" digit-len="6" inter-digit-timeout="1000" max-failures="3"> <entry action="menu-exec-app" digits="*" param="abc voicemail"/> </menu> -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

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Send Remote-party-I to Cisco CUCM SIP trunk
by Yehavi Bourvine 10 Mar '20

10 Mar '20

Hello, I have a setup where Freeswitch is a transit PBX between the public world and two internal PBXes - one is another Freeswitch and one is Cisco CUCM. All connections are SIP trunks. The issue is that the external world SIP trunk sends the real called party name in the "200 OK" in the P-Asserted-Identity field. At this stage, the "transit FS" sends also OK without the received called party name, and then the behavior of the two internal PBXes is different: - When the internal PBX is FS, the transit FS generates INFO message and places the updated called name in X-FS-Display-Name field. - When the internal PBX is Cisco, nothing is sent, thus the true called name is not displayed. Any idea how to force the transit FS to send it to Cisco as well? Thanks, __yehavi:

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can not use freeswitch.EventConsumer
by Babak Yakhchali 10 Mar '20

10 Mar '20

Hi I'm trying to use a simple lua event consumer like this: *connnn = freeswitch.EventConsumer('CHANNEL_HANGUP');* freeswitch.consoleLog("NOTICE", "waiting for event \n"); e = con:pop(1,500) freeswitch.consoleLog("NOTICE", "ssssssssssssssstop waiting for event \n"); freeswitch.consoleLog("NOTICE", e:serialize().."\n"); but lines after consumer creation are never executed! if I comment the first line, next lines are executed successfully, scripts blocks at the first line my lua xml config is like this: <configuration name="lua.conf" description="LUA Configuration"> <settings> <param name="module-directory" value="/usr/local/lib/lua/5.2/?.so"/> <param name="startup-script" value="jobs/rk-token-update.lua"/> <hook event="DETECTED_TONE" script="on_tone.lua"/> <hook event="CHANNEL_BRIDGE" script="onbridge.lua"/> hook event="CHANNEL_UNBRIDGE" script="onbridge.lua"/> </settings> </configuration> fs is an old box with version 1.6 FreeSWITCH Version 1.6.20+git~20180507T185611Z~43a9feb7f8~64bit (git 43a9feb 2018-05-07 18:56:11Z 64bit)

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Building for Redhat Linux
by Larry Hemenway 09 Mar '20

09 Mar '20

All, I'm trying to build from source using the instructions here: https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 I am almost there but continuously fail to build mod_av with the following error: make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_av' Makefile:1382: *** You must install libavformat-dev to build mod_av. Stop. I've tried building libav from scratch from their github repository and still see the same error. Has anyone else had success with this? Larry -- *Your privacy is important to us. That is why we have taken appropriate measures to ensure the data you provide to us is kept secure. To learn more about how we process your personal information, how we comply with applicable data protection laws, and care for the security and privacy of your personal data, please review our Privacy Policy. If you have any questions related to data protection and compliance with applicable laws, please contact us at our Security Operations Center at 1-800-674-4357 or mail us at: Attention: Privacy Compliance Program, P.O. Box 59263, Schaumburg, IL USA, 60159-0263*

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