On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis <jerry.g...@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@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@IP:5060>;tag=IP+3+67d18b6f+9e6ad02d^M To: <sip:Called-Number@dnsname>;tag=as128621a0^M Call-ID: 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@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
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