[asterisk-users] Faking RTP
Hi, I am using the ICES application and one issue we are having is the carrier is timing out because we are not sending RTP. I did try RTP keepalive and that did not help. Is anyone aware of a way to have Asterisk send a fake RTP packet (as in a real RTP packet with no audio) in place of RTCP packets? TIA. Regards, Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing audio and video to linux
Is it possible to have a video call running H264 for example and somehow pass that to linux through pulseaudio and also the video to /dev/video or something ? such that some other program can have access - like gstreamer? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E-Mail notification for each received call
Hi everybody, we use Asterisk to route all calls to a inbound phone number to a specific outbund mobile phone number, depending on time and date. I'd like to send a notification email to a specific email address, each time we receive a call. For this I used the tip of "dicko" here [1]. I'm a Asterisk newbie. Unfortunately it doesn't work. The System() command is not executed. I've tried to execute the a simple bash script for testing purposes (write a test string with > into a file, it's attached) - even that doesn't work. Even if I try it with System(/bin/bash ), it makes no difference. My test bash script and extension_custom.conf file is attached. This advice, which is is related to the first one, didn't help, too. [2] Our system is the most current FreePBX Distro release, Asterisk 16.9.0 and DAHDI. (We can't use VoIP, the communications department uses a proprietary protocol.) Also "core set debug"/"core set verbose" didn't help. Any help is appreciated. Kind regards, Kai [1] https://community.freepbx.org/t/send-email-or-sms-notification-for-every-inbound-hangup-call/45169 [2] https://community.freepbx.org/t/email-notification-of-incoming-missed-call/29913 #!/bin/bash cd ~asterisk echo test > /home/asterisk/test [macro-hangupcall-custom] exten => s,1,DumpChan() exten => s,n,System(/home/asterisk/bash_test) exten => s,n,MacroExit() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] Re: Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC
Hello, Adding tthe following lines in modules.conf made VoiceMailMain re-appear. load = app_voicemail.so noload = app_voicemail_odbc.so noload = app_voicemail_imap.so My previous modules.conf only included: load = app_voicemail.so I hope this would help others. Best regards Le mer. 25 mars 2020 à 15:11, Olivier a écrit : > Hello, > > On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. > > I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using > classical File module (in modules;conf and voicemail.conf): > cd asterisk-17.3.0 > ... > make menuselect.makeopts > menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; > done > menuselect/menuselect --enable app_voicemail_odbc menuselect.makeopts; > done > ... > > I've got this: > > CLI> core show application VoiceMailMain > Your application(s) is (are) not registered > Command 'core show application VoiceMailMain' failed. > > CLI> module reload app_voicemail.so > Module 'app_voicemail.so' reloaded successfully. > -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail > System)) > > CLI> module show like app_voicemail.so > Module Description > Use Count Status Support Level > app_voicemail.so Comedian Mail (Voicemail System) 0 > Running core > 1 modules loaded > > > Then I re-compiled from source removing both app_voicemail_imap and > app_voicemail_odbc menuselect options and I could successfully get : > > CLI> core show application VoiceMailMain > > -= Info about application 'VoiceMailMain' =- > ... > > > What are the necessary steps to have app_voicemail, app_voicemail_imap and > app_voicemail_odbc available along with VoiceMailMain when still using > app_voicemail as preferred Voicemail module ? > > Best regards > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 09:18:24 -0400, Doug Lytle wrote: > > >>> Can I adjust the talk or listen volume for another user? > > I've never used the volume controls, but it would appear. > > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration > > Doug According to this document, there is no way for me to change the volume(s) for another user, whereas meetme allows me to do this by specifying the conference number and user number. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
>>> Can I adjust the talk or listen volume for another user? I've never used the volume controls, but it would appear. https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 06:54:37 -0400, Doug Lytle wrote: > > >>> I never moved to confbridge because they don't have an option for > >>> controlling the volume of other > >>> participants audio > > I have menu options in my confbridge configs that has increase and decrease > conference volume. > > I'd still configure a small confbridge and test if you still have the issue, > since meetme is no longer being developed. Can I adjust the talk or listen volume for another user? If I could do that I would switch, but otherwise I have to stay with meetme. And I wonder if its a meetme issue? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
>>> I never moved to confbridge because they don't have an option for >>> controlling the volume of other >>> participants audio I have menu options in my confbridge configs that has increase and decrease conference volume. I'd still configure a small confbridge and test if you still have the issue, since meetme is no longer being developed. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
