> On 16 Sep 2017, at 01:56, Phil Mickelson <phil@xxxxxxxxxxxxxxx> wrote: > > Sorry, I wish I could help you further but I'm not that familiar with pjsip. Perhaps someone from Digium will respond. > > On Fri, Sep 15, 2017 at 12:46 PM, Richard Frith-Macdonald <richard.frith-macdonald@xxxxxxxxxxxxxxxx> wrote: > > > On 15 Sep 2017, at 13:09, Richard Frith-Macdonald <richard.frith-macdonald@xxxxxxxxxxxxxxxx> wrote: > > > > > >> On 15 Sep 2017, at 12:19, Phil Mickelson <phil@xxxxxxxxxxxxxxx> wrote: > >> > >> If you look at the Channel option (https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Channels+REST+API#Asterisk15ChannelsRESTAPI-create) you see the originateWithId POST option. I believe that's what you want. And then you specify the caller id you want to use in the callerId query parameter. It works like a champ. I can change this easily for what ever customer I'm making the call for. > >> > >> Hope this helps. If not, let me know and I'll try again. > > > > Thanks. I'm using Asterisk 14.6 rather than 15 at the moment, but I don't think there should be a difference (I guess I need to carefully read the new features in 15 though), > > > > I'm trying to use the /channels/create command rather than the /channels command, because I want to be able to bridge the call before the dial-out. > > > > I'll try using /channels instead (if it works it would at least prove that the problem is not in the endpoint configuration), but in the long run I thing I need to be able to do the job using /channels/create > > If I use /channels/{newChannelId} rather than /channels/create asterisk *does* put the value from the 'callerId' parameter into the 'From' header of the INVITE (though not into the Contacts header). > I don't know enough about SIP to know if that's sufficient for my callerid to show up at the remote phone, but it looks promising. > Anyway, that success seems to rule out an error in the pjsip endpoint configuration. > > I guess setting the CALLERID(num) variable on a channel created using /channels/create is not sufficient to get it sent, but I don't see what else I might need to do. Thanks ... with the clue that /channels/{newChannelId} works, I was able to go through the asterisk source and find out what it did with the callerId parameter that was different from setting CALLERID(num). It turns out that chanel callerId is specified it effectrively sets CALLERID(num), CALLERID(num_valid), CONNECTEDLINE(num) and CONNECTEDLINE(num_valid). And it seems that, in order to get the value set in the SIP INVITE's From header, the important thing to set is actually CONNECTEDLINE(num) _______________________________________________ asterisk-app-dev mailing list asterisk-app-dev@xxxxxxxxxxxxxxxx http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
