Re: 401 error
"Joshua C. Colp" <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CAM0A2Z3V_yttYCxJxpn1e-6_crseDNdObiLBY7bV-571d4fc4A@mail.gmail.com> |
On Thu, Mar 9, 2023 at 10:43 PM Jerry Geis <[email protected]> wrote: > I have a SIP trunk - calls going out work fine. > > Trying to setup an incoming call with a DNIS > > When I dial the number - I see nothing on the CLI. > The person says the server is returning 401 > > How do I debug that. Using asterisk 18.8.0 > There are two different SIP channel drivers. If using chan_sip then "sip set debug on" will show you the SIP traffic, if using chan_pjsip then "pjsip set logger on" will. After confirming it you then look at the configuration. You would need to ensure that you are matching the incoming traffic against either a peer for chan_sip (host= in a peer), or an endpoint in chan_pjsip (identify section). You'd also need to confirm that you haven't configured it to challenge those calls for authentication (insecure=very in chan_sip, and not having auth or inbound_auth set on endpoint in chan_pjsip). -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _____________________________________________________________________ -- 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