[asterisk-users] SIP credentials in the dialplan

[Joshua C. Colp]

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