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 Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230309/4c5d5979/attachment.html>
On Thu, 9 Mar 2023, Jerry Geis wrote:> Trying to setup an incoming call with a DNIS > > When I dial the number - I see nothing on the CLI.Have you enabled [PJ]SIP debugging? Bumping up console debug and verbose levels may also yield clues. tcpdump+sngrep are my 'gotos' for packet analysis, but this may not need too much depth.> The person says the server is returning 401 > > How do I debug that. Using asterisk 18.8.0https://en.wikipedia.org/wiki/List_of_SIP_response_codes#:~:text=401%20Unauthorized,1%5D%3A%E2%80%8A%C2%A721.4.2 "401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars.[1]: §21.4.2" My guess would be a user or password mismatch. Are you using SIP or PJSIP? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281
On Thu, Mar 9, 2023 at 10:43 PM Jerry Geis <jerry.geis at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230310/a4a4514c/attachment.html>
On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis <jerry.geis at gmail.com> 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 > > Thanks > > Jerry >Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. Using INVITE request as basis request - 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP at IP Found peer 'JJ' for 'phone' from IP:5060 <--- Reliably Transmitting (no NAT) to IP:5060 ---> SIP/2.0 401 Unauthorized^M Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M From: "Caller" <sip:phone at IP:5060>;tag=IP+3+67d18b6f+9e6ad02d^M To: <sip:Called-Number at dnsname>;tag=as128621a0^M Call-ID: 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP at IP ^M CSeq: 503124310 INVITE^M Server: Asterisk PBX 18.14.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE^M Supported: replaces, timer^M WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cbb5c2f"^M Content-Length: 0^M I dont see a reason why it failed. I tried nat=yes, made no difference. I tried insecure=very, made no difference. I do have: externip=X localnet=Y localnet=Z set in sip.conf As I mentioned - I can call out over this SIP trunk. What next ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230310/4122d5b3/attachment.html>