On Fri, Mar 10, 2023 at 9:49 AM Jerry Geis <jerry.g...@gmail.com> wrote:
> > > 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 > Just added insecure=very again, stopped and started. [JJ] type=friend dtmfmode=rfc2833 secret=yes username=NUMBER defaultuser=NUMBER disallow=all allow=ulaw allow=alaw context=smvoice-incoming host=dnsname canreinvite=yes qualify=yes insecure=very Got the same 401. Thanks Jerry
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