March 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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webRTC on local network
by Francesco Facco de Lagarda 07 Mar '20

07 Mar '20

Good morning to all. I am developing a pabx and video conferencing system to be used inside Italy’s Penal Institutions. Because of very stringent security regulations, no access is granted to external Internet. The server is an FS 1.6 The clients are Raspberry PI based video terminals running and adapted version of the Verto demo client. Can anyone please give me some functioning pointers on how I can get around the age-old problem of certificates, etc on an internal network. I’ve dredged the ‘net, but have never found a satisfying answer. Thank you in advance for any assistance, Francesco

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[Stream Media To Other systems]
by Bilal Abbasi 06 Mar '20

06 Mar '20

Hi Users, I am checking if someone has streamed the media out of FreeSWITCH to other system for analysis purposes. I want call media to be sent out to some other systems for analysis. Currently i tried to use mod_vlc but getting ERR: VLC: PulseAudio server connection failure: Connection refused I did checked that pulse audio is running on system, and could not see any related answer for that error online. Can someone please help me in a direction, or guide me better module/strategy to be used here. Regards Abbasi

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mod_perl does not compile under buster
by John Covici 06 Mar '20

06 Mar '20

Hi. I am trying to set up fs 1.10.2 under Debian buster, but mod_perl will not compile. I get the following error: make[4]: Entering directory '/usr/src/freeswitch/src/mod/languages/mod_perl' CC mod_perl_la-mod_perl.lo CXX mod_perl_la-freeswitch_perl.lo CXX mod_perl_la-mod_perl_wrap.lo CC mod_perl_la-perlxsi.lo CXXLD mod_perl.la /usr/bin/ld: cannot find -lgdbm_compat collect2: error: ld returned 1 exit status Now here are the gdbm packages I have: ii libgdbm-compat4:amd64 1.18.1-4 amd64 GNU dbm database routines (legacy support runtime version) ii libgdbm-dev:amd64 1.18.1-4 amd64 GNU dbm database routines (development files) ii libgdbm3:amd64 1.8.3-14 amd64 GNU dbm database routines (runtime version) ii libgdbm6:amd64 1.18.1-4 amd64 GNU dbm database routines (runtime version) Am I missing a package or otherwise doing something wrong? 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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Re: [Freeswitch-users] LUA Scripting - When to check session readiness?
by Giovanni Maruzzelli 06 Mar '20

06 Mar '20

clutter the code the most you can bear, and *definitely* before *ANY* blocking operation (operation that wait for an answer. Maybe you don't know that operation wait for an answer. Clutter the code ) On Thu, Mar 5, 2020 at 10:12 PM Wilmar Pérez via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "Wilmar Pérez" <ewdpb(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 5 Mar 2020 09:15:37 -0500 > Subject: LUA Scripting - When to check session readiness? > Hi all, > > I am coding and IVR on LUA. Based on the freeswitch books you have to > check session readiness to avoid getting stack in a nasty error or infinite > loop. That is OK. My question is: when should I do it? What I am doing now > is to check it before asking caller for feedback or querying a Web > Service. Does it make sense? It bothers me that the code gets really > cluttered b the constant checking (LUA code layout does not help either). > > I am very aware this is not exact science, I just want to know the > opinion of more experienced people out there. > > Thanks! > > Wilmar > > > > ---------- Forwarded message ---------- > From: "Wilmar Pérez via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 05 Mar 2020 13:12:31 -0800 (PST) > Subject: [Freeswitch-users] LUA Scripting - When to check session > readiness? > _________________________________________________________________________ > > 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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Web socket overload
by Hưởng Ngô 06 Mar '20

06 Mar '20

Hi, I'm using freeswitch latest version 1.10.2 release 14. Now, I'm using webRTC, in this internal.xml file, I add param port 7446 for webRTC, I have for 900 TCP connection to socket but It is overload, extension not register to socket When I restart freeswitch service, It need for 5-10 minutes to free socket connection, and extension register to socket success. I debug for 3 days, but not detect problem. Can you help me with this problem? Thank you!

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Re: [Freeswitch-users] Custom audio files auto bitrate
by Dragos Oancea 05 Mar '20

05 Mar '20

You mean sample rate. Not sure, but maybe these are useful: https://freeswitch.org/confluence/display/FREESWITCH/mod_local_stream https://freeswitch.org/confluence/display/FREESWITCH/Music+on+Hold Or you could check in the dialplan channel variable "rtp_use_codec_rate" and play different files based on its value. On Thu, Mar 5, 2020 at 7:58 PM Wilmar Pérez via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "Wilmar Pérez" <ewdpb(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 5 Mar 2020 11:09:58 -0500 > Subject: Custom audio files auto bitrate > Hi all, > > I have a bunch of custom audio files to playback in an IVR execution. How > can I make it work as the default sound files in freeswitch where the audio > rate is auto selected (8000, 16000, 32000, 48000)? > > Thanks! > > Wilmar > > > > ---------- Forwarded message ---------- > From: "Wilmar Pérez via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 05 Mar 2020 11:58:52 -0800 (PST) > Subject: [Freeswitch-users] Custom audio files auto bitrate > _________________________________________________________________________ > > 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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