search for: 13778ac6c6ac51ac853b3fb4a387e5b71

...t. 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...

...s 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...

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: