[asterisk-users] Bare Metal vs Containers/vms
Hi All, I vaguely remember someone at Astricon making the case for having multiple containers/vps each running asterisk vs using asterisk direct on bare metal. Something about getting better performance. Does anyone have any insight on this? TIA and stay safe Dovid PS I know vps != containers I just don’t recall if the argument was for vps, containers or both instead of installing direct on bare metal. -- _ -- 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] SIP TLS not working, Asterisk 16.9.0
Hi Stefan, thanks a lot. It is working now. Best regards, Karsten Am Freitag, den 01.05.2020, 18:40 +0200 schrieb Stefan Tichy: > Hi Karsten, > > > On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > > > > The server sends Server Hello, Certificate, Server Key > > Exchange and Server Hello Done. > Something in that packet seems to be unacceptable for openssl 1.1.1d > as it is compiled and configured for Buster. > > Certificate length, Digest algorithm, ... > > > You my change the system default settings at the bottom of > "/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in > mind that this will affect the whole server. > > > > > -- > Stefan Tichy ( asterisk3 at pi4tel dot de ) > -- _ -- 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] SIP TLS not working, Asterisk 16.9.0
Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the bottom of "/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in mind that this will affect the whole server. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- 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] Length of dial string
Or you could just increase MAX_EXTENSION and recompile. On Fri, 01 May 2020 06:25:36 -0400, Paddy Grice wrote: > > [1 ] > [1.1 ] > Hi Dovid > > Yes was one of the options but as the required list is dynamic becomes very > messy - and all combinations problem - where as "call all workers job xxx" > is what is needed so the ability to call 20+ numbers is what is needed - agi > does a database search for all jobx workers and constructs a dialstring with > SIP, DAHDI and Local devices. > > Can someone tell me where to find maximum string length for the dial data in > the DIAL command > > Paddy > > _ > > From: Dovid Bender [mailto:do...@telecurve.com] > Sent: 01 May 2020 10:26 > To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Length of dial string > > > Paddy, > > Why not use local extensions? You can do something like this. > Exten => > s,1,Dial(Local/set1@call_all&Local/set2@call_all&Local/set3@call_all) > > [call_all] > Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 > Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 > Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 > > > On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote: > > > Hi all > > as per the new release notice for 13.33.0 received today - can anyone advise > me the max limit of the string to the Dial Command - see > * [ASTERISK-27946 > https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - > dial (API): Storage of dialed target uses AST_MAX_EXTENSION > when it shouldn't > > I have been fighting with this issue for months trying to find a solution I > need to call 20+ devices at the same time so dial strings are very long I > cant really use a queue(ringall) which was my original idea as the customer > needs different groups for virtually every call some of which are simple sip > devices and others have to be local devices (Internal and External CLIs). > > Paddy Grice > > > > > > -- > _ > -- 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 > > [1.2 ] > [2 ] > -- > _ > -- 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 -- 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] Length of dial string
Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum string length for the dial data in the DIAL command Paddy _ From: Dovid Bender [mailto:do...@telecurve.com] Sent: 01 May 2020 10:26 To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Length of dial string Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1@call_all&Local/set2@call_all&Local/set3@call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote: Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I need to call 20+ devices at the same time so dial strings are very long I cant really use a queue(ringall) which was my original idea as the customer needs different groups for virtually every call some of which are simple sip devices and others have to be local devices (Internal and External CLIs). Paddy Grice -- _ -- 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 -- _ -- 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] Mute conference participants
Doug, I am working with a school where they want the students to be able to talk to the students but sometimes they don't want to allow the students to mute themselves. It seems as if you give power to unmute you can't stop it at all. On Sun, Apr 26, 2020 at 3:09 PM Doug Lytle wrote: > On 4/26/20 10:48 AM, Dovid Bender wrote: > > Hi, > > > > Looking at > > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there > > > is an option for admin_toggle_mute_participants however the non admin > > users can still toggle toggle_mute. Is there any option for the admin > > to disallow non admins from using toggle_mute to unmute themselves? If > > there isn't such an option on there any devs here that can ping me off > > line what it would cost/take to get it done? > > > > > > > Dovid, > > My guess would be to redefine their menu map and take away the option > completely, > > 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 -- _ -- 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] Length of dial string
Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1@call_all&Local/set2@call_all &Local/set3@call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote: > Hi all > > as per the new release notice for 13.33.0 received today - can anyone > advise > me the max limit of the string to the Dial Command - see > * [ASTERISK-27946 > https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - > dial (API): Storage of dialed target uses AST_MAX_EXTENSION > when it shouldn't > > I have been fighting with this issue for months trying to find a solution I > need to call 20+ devices at the same time so dial strings are very long I > cant really use a queue(ringall) which was my original idea as the customer > needs different groups for virtually every call some of which are simple > sip > devices and others have to be local devices (Internal and External CLIs). > > Paddy Grice > > > > > > -- > _ > -- 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 -- _ -- 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] Webrtc and iOS devices
Hello, I upgraded to 16.9.0 and then 16.10.0. I feel that there is something in iOS webrtc implementation which cause issues. If womebody has succeeded with combination iOS, browser (like Safari) and webrtc (conference call), it would be nice to hear. Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 13.41: I honestly couldn't tell you if it would resolve it but there aren't many people going to be willing to help problem solve anything if you're running 13 - you'll get more support on 17 for example. Very easy to bring up a new instance or VM in the grand scheme of things to test the theory and get it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM Teijo wrote: Hello, Currently audio conference. Should upgrading Asterisk from 13 to newer version resolve webrtc/iOS problem? Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 12.18: First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo wrote: Hello, Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, because several other browsers from Windows seem to work. Best regards, Teijo -- _ -- 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 -- _ -- 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 -- _ -- 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] Length of dial string
Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I need to call 20+ devices at the same time so dial strings are very long I cant really use a queue(ringall) which was my original idea as the customer needs different groups for virtually every call some of which are simple sip devices and others have to be local devices (Internal and External CLIs). Paddy Grice -- _ -- 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
