[asterisk-users] 401 error

Jerry Geis jerry.geis at gmail.com
Fri Mar 10 08:49:40 CST 2023


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
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