401 Error

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Asterisk Users 6 Comments

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

6 thoughts on - 401 Error

  • 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).

  • 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” ;tag=IP+3+67d18b6f+9e6ad02d^M
    To: ;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

  • It matched peer ‘JJ’. That peer would need to have insecure=very set, and chan_sip then reloaded. Providing the actual peer would also be faster for anyone to provide help.

  • 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

  • That’s the extent of my vague memories of chan_sip then, someone else may be able to answer.

  • Thank you for the suggestions – it got me to this insecure=port,invite

    This worked.

    Jerry