On 17.02.21 at 21:46 Luca Bertoncello wrote: > Am 16.02.2021 um 22:32 schrieb Michael Maier: > > Hi Michael > >>> Maybe could you send me an abstract of your configuration? >> >> Take a look here [1] > > So, maybe I got it... > I tested the configuration with my Fax number and it seems to work (= I > can call the fax and can call my mobile phone from the fax with > "originate...").
Congrats! > On the registration I have: > > [pbxfax] > type = registration > retry_interval = 20 > max_retries = 10 > contact_user = 00493514977291 > expiration = 120 > transport = transport-udp > outbound_auth = pbxfax > client_uri = sip:03514977...@tel.t-online.de > server_uri = sip:tel.t-online.de > > First: can I use tel.t-online.de or _MUST_ I change it? No, you mustn't change it. You must use tel.t-online.de. > If I understand > your previous E-Mail, I'd say that I can leave tel.t-online.de... Correctly! > Then I have a question by the Dialplan... Currently I have: > > [fax-out] > exten => _X.,1,NoOp() > exten => _X.,n,Verbose(2,Call from FAX) > exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R) > > And I'll replace it with: > > [fax-out] > exten => _X.,1,NoOp() > exten => _X.,n,Verbose(2,Call from FAX) > exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R) > > Is it correct? I tried with > "PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.de,,R" and it does NOT work... > Is it correct, that I have to leave "sip:..."? Don't know - I don't care about dialplan - I'm using FreePBX :-) Thanks Michael -- _____________________________________________________________________ -- 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