asterisk users - Jun 2020 - Forbidden call

I have a call from a call file:

Action: Originate
Async: yes
Channel: SIP/2012
Codecs: ulaw,alaw,gsm
Context: dialout
Exten: callprogress
Priority: 1
Timeout: 20000
Variable: SIPADDHEADER="Alert-Info: Ring Answer"
ActionID: 100014
CallerID: Axis < 525 >


The SIP/2012 is a IP Speaker on the computer. The error is:
[Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite:
Received response: "Forbidden"

Why am I getting "Forbidden" ? Its a call file on my server and the
speaker
is directly connected to my server.

Thanks

Jerry
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On Thu, 11 Jun 2020, Jerry Geis wrote:
> I have a call from a call file:
This looks a lot more like an AMI event than a call file. In any case, it 
doesn't matter.
> Action: Originate
> Async: yes
> Channel: SIP/2012
> Codecs: ulaw,alaw,gsm
> Context: dialout
> Exten: callprogress
> Priority: 1
> Timeout: 20000
> Variable: SIPADDHEADER="Alert-Info: Ring Answer"
> ActionID: 100014
> CallerID: Axis < 525 >
> The SIP/2012 is a IP Speaker on the computer. The error is:
> [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite:
Received response: "Forbidden" 
> 
> Why am I getting "Forbidden" ? Its a call file on my server
It's not a call file permissions thing. That would be a different error 
and reported by something before chan_sip.
> the speaker is directly connected to my server.
How is an IP speaker 'directly connected?' Do you mean directly from the
Ethernet on the speaker to a NIC on the computer? It doesn't matter, just 
curious :)

The only thing that will tell you what is going on is the packets. Crank 
up 'sip set debug on' and see if that yields a clue.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281

Hi Steve, - Your right - the file was AMI (copied the other one).  By
direct connect I simply meant the speaker is an extension on that server.

here is the SIP debug
<--- SIP read from UDP:X.X.X.X:1024 --->


  == Using SIP RTP CoS mark 5
Audio is at 16060
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to X.X.X.X :1024:
INVITE sip:2012@ X.X.X.X :1024;ob SIP/2.0
Via: SIP/2.0/UDP X.X.X.X :5060;branch=z9hG4bK2555a6ef;rport
Max-Forwards: 70
From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66
To: <sip:2012@ X.X.X.X :1024;ob>
Contact: <sip:XXXXXXXXXX@ X.X.X.X :5060>
Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.33.0
Date: Fri, 12 Jun 2020 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Alert-Info: Ring Answer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1889524876 1889524876 IN IP4 X.X.X.X
s=Asterisk PBX 13.33.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 16060 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called 2012

<--- SIP read from UDP: X.X.X.X :1024 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X :5060;rport=5060;received= X.X.X.X
;branch=z9hG4bK2555a6ef
Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060
From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66
To: <sip:2012@ X.X.X.X ;ob>
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP: X.X.X.X :1024 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP X.X.X.X :5060;rport=5060;received= X.X.X.X
;branch=z9hG4bK2555a6ef
Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060
From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66
To: <sip:2012@ X.X.X.X ;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to X.X.X.X :1024:
ACK sip:2012@ X.X.X.X :1024;ob SIP/2.0
Via: SIP/2.0/UDP X.X.X.X :5060;branch=z9hG4bK2555a6ef;rport
Max-Forwards: 70
From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66
To: <sip:2012@ X.X.X.X :1024;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI
Contact: <sip:XXXXXXXXXX@ X.X.X.X :5060>
Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.33.0
Content-Length: 0


---
[Jun 12 08:18:18] WARNING[12933]: chan_sip.c:24191 handle_response_invite:
Received response: "Forbidden" from '"Jerry Geis 101"
<sip:XXXXXXXXXX@
X.X.X.X >;tag=as5e61ec66'
Scheduling destruction of SIP dialog '361b4b803f214946320c0af84a9ac0c4@
X.X.X.X :5060' in 32000 ms (Method: INVITE)
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On Fri, 12 Jun 2020, Jerry Geis wrote:

Any chance you can configure the speaker to syslog to your host so you may 
get a clue why the speaker is rejecting?

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281