Actually SIPFROMDOMAIN was documented here: https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables , but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton <rnew...@digium.com> wrote: > On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson > <torbjorn.abrahams...@gmail.com> wrote: >> I have a problem where I would like to be able to send an arbitrary SIP >> domain when sending a call to a registered friend. By default the from >> domain is set to the IP of the Asterisk server, but I would like to set it >> to something else. >> The case is that when a call from a foreign domain comes in to the Asterisk, >> it will connect it to the callee (but with the domain changed). When the >> callee wants to make a redial from call history, the domain will not be >> correct. >> I could probably do something with the fromdomain setting of the friend, but >> I would like it to be dynamic, ie not having to update the friend definition >> every time a different domain is used. >> I understand that I would need to use outbound proxy in the client to >> prevent it from dialing the domain directly. >> Is this possible? Any alternatives? > > I'm a little confused about what you want to do, however I'll throw > some information at you in hopes that it will help out. > > I did a little research and found that you can set the outbound From > header domain and From header user through two channel variables: > SIPFROMDOMAIN, SIPFROMUSER > > They are sparsely documented, but there is an example in extensions.conf > > same => n(from),Set(__SIPFROMUSER=${CALLERID(num)}) > same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) > ; check if we set the FREENUMDOMAIN global variable in [global] > same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) > ; if we did set it, then we'll use it for our outbound dialing > domain > > It looks like they were added in 1.6.2.X of Asterisk, so if you are > using 1.8.X or above, you should have them. > > On your inbound call, you could use the function SIP_HEADER[1] to > gather the domain and store it for later use when you want to set it > on the outbound call. Though I'm not sure how you could tell that the > call was a redial. > > [1]: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER > > I'm assuming when your SIP client redials that it calls through > Asterisk and is not dialing the previously caller directly. > > Hope any of that helps. *Goes off to document SIPFROMDOMAIN and > SIPFROMUSER on the wiki* > > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > Check us out at: http://digium.com & http://asterisk.org -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users