[asterisk-users] Mixing PJSIP realtime and flat files
Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth, and context columns set but all other fields NULL. Then the end-customer could add a [asterisk-1] section in pjsip.conf which sets the codecs they want to enable. We tried this but it seemed that the [asterisk-1] section in pjsip.conf had no effect. Our sorcery.conf is attached. Is this possible, and how do we do it? Thanks very much for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 sorcery.conf Description: Binary data -- _ -- 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] Mixing PJSIP realtime and flat files
David Cunningham wrote: Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth, and context columns set but all other fields NULL. Then the end-customer could add a [asterisk-1] section in pjsip.conf which sets the codecs they want to enable. We tried this but it seemed that the [asterisk-1] section in pjsip.conf had no effect. Our sorcery.conf is attached. Is this possible, and how do we do it? Thanks very much for any advice. It's not possible to do this. Each source (realtime, config file) provides the complete definition. -- 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] Mixing PJSIP realtime and flat files
Shame, but thank you very much for the reply Joshua. On 22 January 2016 at 10:26, Joshua Colpwrote: > David Cunningham wrote: > >> Hello, >> >> Is it possible to mix PJSIP realtime and flat file configuration in >> pjsip,conf? >> >> What we want is to set up endpoints in the ps_endpoints table with some >> columns set but most being NULL, and then allow end-customers to >> optionally add configuration by adding a pjsip.conf section. >> >> For example, in ps_endpoinds might be an endpoint with id "asterisk-1" >> with the transport, aors, auth, and context columns set but all other >> fields NULL. Then the end-customer could add a [asterisk-1] section in >> pjsip.conf which sets the codecs they want to enable. >> >> We tried this but it seemed that the [asterisk-1] section in pjsip.conf >> had no effect. Our sorcery.conf is attached. >> >> Is this possible, and how do we do it? Thanks very much for any advice. >> > > It's not possible to do this. Each source (realtime, config file) provides > the complete definition. > > -- > 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 > -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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