Otdfbt

I am trying to figure out why my FXO adapter has suddenly stopped working, it has been a while since it was first configured, I had only changed an internal call timeout setting on the FXO adapter and it suddenly stopped acceping incoming calls to the ring group. I've also been unsuccesful in creating accounts on the Asterisk and freepbx forums but I digress.
I've tried rebuilding the trunk, extension and user associated with the device without any success.

When I make the inbound call I get the following error

 [2016-03-02 12:47:30] ERROR[4687]: res_pjsip.c:2370 sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous'

However it shouldn't be interfacing with PJSIP. My CHAN_SIP bind port is 5061 and the FXO port has been configured to unconditionally call fordward to

601@10.0.100.8:5061

User 601 is a ring group which still works internally. PJSIP is configured to listen on port 5099. (changed to try and prevent it picking up the FXO call)

I've gone from error 401, 500 and all sort of other issues trying to diagnose the problem, days of searching and changing settings hasn't helped yet.

On the Asterisk side, the FXO port is configuued as a trunk, with the following

incoming settings

USER conext=incoming

type=peer
username=60
fromuser=60
insecure=port,invite
host=10.0.100.24
dtmf=rfc2833
port=5062

allow=alaw&ulaw&g729
qualify=yes

This shows up as a peer, but not in the registry. I used to have an extension with the same username but have since deleted it, the username was handling the voicemail on no answer, I'll look at adding it once the call goes throuhg again. The user was configured as PJSIP:600 when it was working, but I've changed it to a new user @ 60 to prevent any old PJSIP configuration from leaking over.

sip show peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
Incoming/60 10.0.100.24 Yes Yes 5062 OK (18 ms)

I'm getting the following from wireshark

SIPStack(1)::cb_rcv: Recieved 403 response for Ttransaction 3(REGISTER)

Which looks like its a FORBIDEN response. Not sure what to check on that however.

If I try to dial the FXO port. I'm getting the following in Wireshark

Call(1)::Call, creating Call Object 1 at port 1:0 with digits <sip:601@10.0.100.8:5061>
ATACtrl::Call, cannot make the call, statusCode = 500, chan status = CALL_DIALED
Dispatching event: 17 (CALL_FAILED)) on port 1:0

and asterisk puts out the error I had at the start of the problem.

[2016-03-02 12:47:30] ERROR[4687]: res_pjsip.c:2370 sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous'

With sip set debug ip 10.0.100.24 I get the following

Reliably Transmitting (NAT) to 10.0.100.24:5062:
OPTIONS sip:10.0.100.24 SIP/2.0
Via: SIP/2.0/UDP 10.0.100.8:5061;branch=z9hG4bK7b2af600;rport
Max-Forwards: 70
From: "Unknown" <sip:6010@10.0.100.8:5061>;tag=as373eb1a0
To: <sip:10.0.100.24>
Contact: <sip:6010@10.0.100.8:5061>
Call-ID: 0e6c3f172fde5de3435f1be434d68911@10.0.100.8:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(13.4.0)
Date: Wed, 02 Mar 2016 02:09:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.100.24:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.100.8:5061;branch=z9hG4bK7b2af600;rport=5061
From: "Unknown" <sip:6010@10.0.100.8:5061>;tag=as373eb1a0
To: <sip:10.0.100.24>;tag=698745166
Call-ID: 0e6c3f172fde5de3435f1be434d68911@10.0.100.8:5061
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0e6c3f172fde5de3435f1be434d68911@10.0.100.8:5061' Method: OPTIONS
[2016-03-02 13:09:42] ERROR[6385]: res_pjsip.c:2370 sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous'

I did find this file, pjsip.endpoint.conf

#include pjsip.endpoint_custom.conf

[anonymous]
type=endpoint
context=from-sip-external
allow=all
transport=udp,tcp,ws,wss

but it should be transporting thorugh normal SIP chanels. I've tried changing ports, creating users and extensions, removing them, always getting the same endpoint for anonymous error.

I'm totaly stumped and would love some ideas on where to look next.

On my system, working with Twilio SIP trunks, this error was resolved by changing the Trunk -> PJSIP Settings -> Registration to None.