On Tue, Jul 9, 2019, at 9:46 AM, Dovid Bender wrote: > > > On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <[email protected]> wrote: > > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > > Hi, > > > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > > should be able to dial with SIP credentials in the DP. Is this still > > > possible in recent versions of Asterisk either with chan_sip or pj_sip? > > > > PJSIP does not currently have functionality to allow such a thing. I > > believe in chan_sip there have been no changes to remove it. > > My DP looks like this: > Exten => aaa,1,Dial(SIP/USERNAME:[email protected]/18005551212) > > > and from the logs I get: > oice1*CLI> console dial aaa@from-external > -- Executing [aaa@from-external:1] Dial("Console/default", > "SIP/USERNAME:[email protected]/18005551212") in new stack > [2019-07-09 08:40:54] NOTICE[27159][C-00019e64]: chan_sip.c:30586 > sip_request_call: Conflicting extension values given. Using 'USERNAME' > and not '1718005551212'
I believe you may want: SIP/1718005551212:password::[email protected] That's at least an example given in the sip.conf.sample file[1], otherwise I'm not sure as I don't have any experience with such Dial lines for chan_sip. [1] https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L51 -- Joshua C. Colp Digium - A Sangoma Company | 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 -- 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
