[asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

[Brian J. Murrell]

I have a trunk set up for the DID from my provider:

[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_username at sip.example.com
retry_interval=60
 
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
 
[my_provider]
type=aor
contact=sip:sip.example.com:5060
 
[my_provider]
type=endpoint
context=from-my_provider
disallow=all
allow=ulaw
outbound_auth=my_provider
aors=my_provider
 
[my_provider]
type=identify
endpoint=my_provider
match=sip.example.com

And it registers fine:

 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 mytrunk/sip:sip.example.com                             my_provider       Registered      


And when it gets an INVITE from my provider (192.168.0.1):

<--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 --->
INVITE sip:1235551212 at 10.75.22.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK06d035fd;rport
Max-Forwards: 70
From: "Fred Flintstone" <sip:4565551212 at 192.168.0.1>;tag=as539f9476
To: <sip:1235551212 at 10.75.22.5:5060>
Contact: <sip:4565551212 at 192.168.0.1:5060>
Call-ID: 3ef877dc4477d8ce4aae29965c5d0875 at 192.168.0.1:5060
CSeq: 102 INVITE
User-Agent: foobar
Date: Sat, 26 Jan 2019 17:40:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Fred Flintstone" <sip:4565551212 at 192.168.0.1>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 295

[SDP redacted]

It logs an error:

[Jan 26 12:40:00] NOTICE[21775]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"Fred Flintstone" <sip:4565551212 at 192.168.0.1>' failed for '192.168.0.1:5060' (callid: 3ef877dc4477d8ce4aae29965c5d0875 at 192.168.0.1:5060) - No matching endpoint found

But then goes on to complete the call:

<--- Transmitting SIP response (352 bytes) to UDP:192.168.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;rport=5060;received=192.168.0.1;branch=z9hG4bK06d035fd
Call-ID: 3ef877dc4477d8ce4aae29965c5d0875 at 192.168.0.1:5060
From: "Fred Flintstone" <sip:4565551212 at 192.168.0.1>;tag=as539f9476
To: <sip:1235551212 at 10.75.22.5>
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.1
Content-Length:  0

[ launch into dialplan ]

So why the spurious error when it was able to complete the call?

Cheers,
b.

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