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

Reply via email to