[asterisk-users] SIP credentials in the dialplan
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? TIA. Dovid -- _ -- 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
Re: [asterisk-users] SIP credentials in the dialplan
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. -- 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
Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/7/19 11:55 AM, Michael Maier wrote: > On 06.07.19 at 22:16 hwilmer wrote: >> Is there an advantage to using pjsip? What's needed for easybell with pjsip? > > For easybell, I don't know of any advantage. But that's not very reliable, > because I'm using easybell for dedicated requirements only. I'm considering > chan_sip legacy. I wouldn't build any new installation on chan_sip (if there > are no technical > contradictions). > > Easybell does have a pretty fine documentation for FreePBX and pjsip: > https://www.easybell.de/nc/hilfe/ergebnis/freepbx-130124-mit-asterisk-13.html That's not for asterisk, and most documentations for asterisk are not for pjsip. > [why encryption?] > >> I consider it a requirement for when employees end up using their mobile >> phones over foreign wireless networks, which is something that would be >> virtually impossible to prevent should the asterisk server be made reachable >> from the outside. > > That's true. But don't forget to encrypt RTP at this point! This must be done > additionally. > BTW: easybell doesn't support full RTP encryption. It's supported for > outgoing calls only > (https://en.easybell.de/nc/help/questions/questions-concerning-the-telephone-connection/answer/does-easybell-support-the-data-encryption-srtp-for-voip.html). They also say that encryption is possible: https://en.easybell.de/nc/help/questions/questions-concerning-the-telephone-connection/answer/can-i-encrypt-the-telephony.html I'll have to see what their support says about this. > > That's an example for an inbound call - there isn't any support for RTP > encryption: [...] That would really suck. It is not acceptable that phone calls over the internet shouldn't be encrypted. > [...] -- _ -- 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
Re: [asterisk-users] SIP credentials in the dialplan
On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp 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:passw...@sip1.myproxy.net/18005551212) and from the logs I get: oice1*CLI> console dial aaa@from-external -- Executing [aaa@from-external:1] Dial("Console/default", "SIP/ USERNAME:passw...@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' -- _ -- 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
Re: [asterisk-users] SIP credentials in the dialplan
On Tue, Jul 9, 2019, at 9:46 AM, Dovid Bender wrote: > > > On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp 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:passw...@sip1.myproxy.net/18005551212) > > > and from the logs I get: > oice1*CLI> console dial aaa@from-external > -- Executing [aaa@from-external:1] Dial("Console/default", > "SIP/USERNAME:passw...@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::usern...@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
Re: [asterisk-users] 302 moved temporally callerid behavior
Lol, everything was too simple. It was just a macro with app Dial with 'f' option configured. Normally I don't use 'f', so I haven't checked that :) вт, 25 июн. 2019 г. в 19:05, Doug Lytle : > core show version > > Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on > 2019-04-05 11:41:43 UTC > > Built from source, > > Douh > > -- > _ > -- 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 -- _ -- 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
Re: [asterisk-users] SIP credentials in the dialplan
Josh, Thanks. I had another look. This seems to work for me: Dial(SIP/18005551212:PASSWORD::usern...@sip1.mydomain.net!! usern...@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 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 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:passw...@sip1.myproxy.net/18005551212) > > > > > > and from the logs I get: > > oice1*CLI> console dial aaa@from-external > > -- Executing [aaa@from-external:1] Dial("Console/default", > > "SIP/USERNAME:passw...@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::usern...@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 -- _ -- 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