[asterisk-users] 401 error
Joshua C. Colp
jcolp at sangoma.com
Fri Mar 10 09:07:00 CST 2023
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.geis at gmail.com> wrote:
>
>
> On Fri, Mar 10, 2023 at 9:49 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 ?
>> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230310/239573a5/attachment.html>
More information about the asterisk-users
mailing list