That's the extent of my vague memories of chan_sip then, someone else may be able to answer.
On Fri, Mar 10, 2023 at 11:05 AM Jerry Geis <jerry.g...@gmail.com> wrote: > > > 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 > > -- > _____________________________________________________________________ > -- 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 -- 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
