asterisk users - Feb 2019 - PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method

I have a PJSIP trunk set up which works fine for voice.  I can call out
and I receive calls from it once it registers.

What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
events.  It was working earlier today but I seem to have done something
as I was enabling voice on the trunk to mess it up.  On receiving of a
MESSAGE, my Asterisk sends a 401 for the ITSP to authenticate it's
message, which it does, to which my Asterisk responds with a "481
Call/Transaction Does Not Exist" and displays nothing at all in the
console.

The configuration of the ITSP is:

[trunk]
type=registration
transport=transport-udp
outbound_auth=trunk-auth
server_uri=sip:itsp.example.com
client_uri=sip:userid at itsp.example.com

[trunk-auth]
type=auth
auth_type=userpass
password=********
username=userid

[trunk-endpoint](!)
type=endpoint
transport=transport-udp
context=from-trunk
message_context=messages
disallow=all
allow=ulaw
from_user=userid
outbound_auth=trunk-auth
auth=trunk-auth
send_pai=yes

[trunk-aor](!)
type=aor
qualify_frequency=15

[trunk-foo](trunk-endpoint)
aors=trunk-foo
[trunk-foo](trunk-aor)
contact=sip:userid at itsp.example.com:5060

[trunk-foo]
type=identify
endpoint=trunk-foo
match=itsp.example.com

The SIP conversation when the ITSP is trying to send the MESSAGE:

<--- Received SIP request (456 bytes) from UDP:10.0.0.1:5060 --->
MESSAGE sip:s at 10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK51d6d4da;rport
Max-Forwards: 70
From: "5555551212" <sip:5555551212 at
itsp.example.com>;tag=as6c34cb69
To: <sip:s at 10.75.22.8:5060>
Contact: <sip:5555551212 at 10.0.0.1:5060>
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
CSeq: 102 MESSAGE
User-Agent: itsp.example.com
X-SMS-To: 5551234567
Content-Type: text/plain;charset=UTF-8
Content-Length: 1

test message
<--- Transmitting SIP response (515 bytes) to UDP:10.0.0.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK51d6d4da
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
From: "5555551212" <sip:5555551212 at
itsp.example.com>;tag=as6c34cb69
To: <sip:s at 10.75.22.8>;tag=z9hG4bK51d6d4da
CSeq: 102 MESSAGE
WWW-Authenticate: Digest
realm="asterisk",nonce="1550441754/[redacted]",opaque="2c504a1035f74a1d",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.25.0
Content-Length:  0


<--- Received SIP request (707 bytes) from UDP:10.0.0.1:5060 --->
MESSAGE sip:s at 10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK516e2ba7;rport
Max-Forwards: 70
From: "5555551212" <sip:5555551212 at
itsp.example.com>;tag=as6c34cb69
To: <sip:s at 10.75.22.8:5060>;tag=z9hG4bK51d6d4da
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
CSeq: 103 MESSAGE
User-Agent: itsp.example.com
Proxy-Authorization: Digest username="userid",
realm="asterisk", algorithm=MD5, uri="sip:s at
172.1.2.3:5060", nonce="1550441754/[redacted]",
response="[redacted]", opaque="2c504a1035f74a1d", qop=auth,
cnonce="544772bb", nc=00000002
X-SMS-To: 5551234567
Content-Type: text/plain;charset=UTF-8
Content-Length: 1

c
<--- Transmitting SIP response (388 bytes) to UDP:10.0.0.1:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP
10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK516e2ba7
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
From: "5555551212" <sip:5555551212 at
itsp.example.com>;tag=as6c34cb69
To: <sip:s at 10.75.22.8>;tag=z9hG4bK51d6d4da
CSeq: 103 MESSAGE
Server: Asterisk PBX 13.25.0
Content-Length:  0

I couldn't find any solutions to a 481 in response to a MESSAGE after
much searching.

I know MESSAGE does work, as I had it working earlier today.  I just
seemed to have messed it up adding the additional configuration to the
trunk to support incoming and outgoing calls.

Any ideas?

Cheers,
b.

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On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell
wrote:
> I have a PJSIP trunk set up which works fine for voice.  I can call
> out
> and I receive calls from it once it registers.
> 
> What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
> events.  It was working earlier today but I seem to have done
> something
> as I was enabling voice on the trunk to mess it up.  On receiving of
> a
> MESSAGE, my Asterisk sends a 401 for the ITSP to authenticate it's
> message, which it does, to which my Asterisk responds with a "481
> Call/Transaction Does Not Exist" and displays nothing at all in the
> console.
Nobody has any idea about this?  :-(

Cheers,
b.

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