On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> 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:PASSWORD at sip1.myproxy.net/18005551212) and from the logs I get: oice1*CLI> console dial aaa at from-external -- Executing [aaa at from-external:1] Dial("Console/default", "SIP/ USERNAME:PASSWORD at sip1.myproxy.net/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' -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190709/dbb4f8ca/attachment.html>
On Tue, Jul 9, 2019, at 9:46 AM, Dovid Bender wrote:> > > On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> 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:PASSWORD at sip1.myproxy.net/18005551212) > > > and from the logs I get: > oice1*CLI> console dial aaa at from-external > -- Executing [aaa at from-external:1] Dial("Console/default", > "SIP/USERNAME:PASSWORD at sip1.myproxy.net/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::username at sip1.myproxy.net 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
Josh, Thanks. I had another look. This seems to work for me: Dial(SIP/18005551212:PASSWORD::USERNAME at sip1.mydomain.net!! USERNAME at sip1.example.net,,) So it seems like I needed to put the called number followed by the password :: and then the username. On Tue, Jul 9, 2019 at 8:57 AM Joshua C. Colp <jcolp at digium.com> wrote:> On Tue, Jul 9, 2019, at 9:46 AM, Dovid Bender wrote: > > > > > > On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> 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:PASSWORD at sip1.myproxy.net/18005551212) > > > > > > and from the logs I get: > > oice1*CLI> console dial aaa at from-external > > -- Executing [aaa at from-external:1] Dial("Console/default", > > "SIP/USERNAME:PASSWORD at sip1.myproxy.net/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::username at sip1.myproxy.net > > 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190709/42900f15/attachment.html>