Re: [asterisk-users] PJSIP defaults for endpoints when using realtime
On Thu, Jul 14, 2016 at 6:45 AM, A J Stileswrote: > On Thursday 14 Jul 2016, Joshua Colp wrote: > > Carlos Chavez wrote: > > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > > (language, dtmf, vmexten, etc) and just leave many fields in the > > > database as NULL. What would be the proper way to do this for Asterisk > > > 13 and PJSIP? > > > > Kia ora, > > > > PJSIP doesn't have the ability in it to override built-in defaults for > > everything. You have to specify it yourself for realtime. If using > > config files then config file templates can be used to do this. > > If the database you are using is MariaDB or MySQL, then you should be able > to > set default values for columns in the table definition. Then when you do > an > INSERT into only some columns, the rest will be populated with the default > values. > > To alter the structure of an already-created table, use something like > > ALTER TABLE stuff CHANGE COLUMN foo foo VARCHAR(20) NOT NULL DEFAULT > "wibble"; > > (Yes, the column name should be there twice: you might want to rename it, > so > you have to specify an old and a new name even if the two are the same.) > > You will then need to use something like > > UPDATE stuff SET foo="wibble" WHERE foo IS NULL; > While this will work, your defaults may not survive the next time alembic is run to upgrade the database. It's rare but we do occasionally drop and re-create columns to change their types. You could also play with insert triggers which are attached to the table instead of specific columns. These might be easier to manage. > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- 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 > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] PJSIP defaults for endpoints when using realtime
On Thursday 14 Jul 2016, Joshua Colp wrote: > Carlos Chavez wrote: > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > (language, dtmf, vmexten, etc) and just leave many fields in the > > database as NULL. What would be the proper way to do this for Asterisk > > 13 and PJSIP? > > Kia ora, > > PJSIP doesn't have the ability in it to override built-in defaults for > everything. You have to specify it yourself for realtime. If using > config files then config file templates can be used to do this. If the database you are using is MariaDB or MySQL, then you should be able to set default values for columns in the table definition. Then when you do an INSERT into only some columns, the rest will be populated with the default values. To alter the structure of an already-created table, use something like ALTER TABLE stuff CHANGE COLUMN foo foo VARCHAR(20) NOT NULL DEFAULT "wibble"; (Yes, the column name should be there twice: you might want to rename it, so you have to specify an old and a new name even if the two are the same.) You will then need to use something like UPDATE stuff SET foo="wibble" WHERE foo IS NULL; -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] PJSIP defaults for endpoints when using realtime
Carlos Chavez wrote: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? Kia ora, PJSIP doesn't have the ability in it to override built-in defaults for everything. You have to specify it yourself for realtime. If using config files then config file templates can be used to do this. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] PJSIP defaults for endpoints when using realtime
with templates. Regards El 13/07/2016 a las 23:49, Carlos Chavez escribió: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- _ -- 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] PJSIP defaults for endpoints when using realtime
Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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