[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
