Re: [asterisk-users] ODBC crashing asterisk
On Mar 24 2016, at 6:01 pm, Mike Diehl mdiehlena...@gmail.com wrote: > Normally, SQL errors don't result in a segfault. I understand that this is a problem with a particular version of the ODBC driver. I just can't find a reference to it at the moment. > > > > Are you tried collect a backtrace? https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC crashing asterisk
March 24 2016 6:01 PM, "Mike Diehl" <mdiehlena...@gmail.com> wrote: > Normally, SQL errors don't result in a segfault. I understand that this is a > problem with a particular version of the ODBC driver. I just can't find a > reference to it at the moment. Are you tried collect a backtrace? https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR records and conferences
On Mar 15 2016, at 8:06 pm, Richard Kenner ken...@gnat.com wrote: > At least in version 12.2.0, the code in cdr.c appears to create CDR records for each pair of users in a conference. This is quadratic and would seem to be an issue with large conferences. > > I got two Asterisk crashes when a lot of people tried to dial into a conference. They appear quite related to > > https://issues.asterisk.org/jira/browse/ASTERISK-24758 If you think is related that issue you can reopen. This issue was closed by not activity, if you put more information and help maybe is possible will resolve in case if was a bug. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WRONG Queues log
March 22 2016 11:44 AM, "Olivier" <oza.4...@gmail.com> wrote: > Which Asterisk version do you use ? > > And what is your exact question ? > What do you expect queue_log file to hold ? I suppose the problem is member SIP/4001 is who answer the call is logged as SIP/4003 in queue_log I really interesting the case but without information like Asterisk version and how can reproduce, i see very hard find or identified the problem -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Panel for app_queue
Hello everyone, I write because recently i released a new version 0.8.0 of QPanel. Qpanel is a dashboard monitor panel for app_queue You can get and try from https://github.com/roramirez/qpanel (Opensource by MIT licence). I appreciate if you can check and would like to hear your comments. You can add new issue (bug, feature or improvement) in https://github.com/roramirez/qpanel/new Regards, -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass variable to voicemail script
March 6 2016 1:06 AM, "Michelle Dupuis" <mdup...@ocg.ca> wrote: > I have a custom voicemail script which reformats and forwards the attached > voicemail wav file to > the recipient. > > I would like to make use of a channel variable in my script; is there a way > to pass a channel > variable to this voicemail script? How you call the custom script? -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 Realtime MusicOnHold
March 4 2016 3:48 PM, "Carlos Chavez" <cur...@telecomabmex.com> wrote: > I am having a problem trying to use the realtime database for > musiconhold for Asterisk 13. Everything is setup and I can see the mapping: > > ===> musiconhold (db=general, table=musiconhold) > > Only what is in the musiconhold.conf file appears in Asterisk and > the database is completely ignored. The database was created using > alembic and the structure included with Asterisk 13.7.2. We are using > MariaDB on CentOS 7 and res_config_mysql for the realtime engine. All > other configurations (pjsip, iax, voicemail and queues) are working as > expected from the database. How can I debug this problem? Apparently is a bug. But if you set in extconfig.conf the musiconhold when in dialplan is used searching in database. The case is not showing when do execute "moh show classe" or "moh show files" You can open a issue in jira https://issues.asterisk.org/jira/browse/ASTERISK Regards, -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash asterisk res_odbc
February 29 2016 5:40 AM, "Maxime" <mcaillet...@openip.fr> wrote: > Hi, > > Thank you for the reply. > > My OS is : Debian 7. > > But i have more than 20 servers with the same features/resources (OS, > material, ... ) without the > issue. > Maybe could debug the crash getting a Backtrace https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace And open a issue on the Jira https://issues.asterisk.org/jira/browse/ASTERISK Regards, -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice recognition IVR Is it possible?
February 23 2016 6:57 PM, "Frank" <mailingl...@linuxista.com> wrote: > On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote: > >> Google?... > > Yeah... searched "google voice recognition api asterisk", browsed though > various results. Nothing helpful for a beginner, very confusing bla > bla... > > Thanks anyway for your help. You can check https://github.com/antirek/voicer I see only today and not tested now. For enable API Key you need do it by Google Console and enabled the service for voice (not remember the name exactly) https://console.developers.google.com Regards, -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windstream SIP Trunk settings
February 23 2016 9:37 AM, "James Cass" <jcas...@gmail.com> wrote: > Thanks everyone, all sound advice. Still can't even get the calls to show up > on the console at all > - I suspect the issue is on the WS side, as I'm not having any issues with > other carriers with > similar settings. You can debug SIP to detect the problem. May be exists some cause tell you more information in the trace SIP. -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
