[asterisk-users] 401 error

[Joshua C. Colp]

On Fri, Mar 10, 2023 at 10:50 AM Jerry Geis <jerry.geis at gmail.com> wrote:

>
>
> On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis <jerry.geis at 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 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 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 ?
>

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.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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